Page 1 Zephyr Xstream Advanced Digital Network Audio Transceiver The Best Way to Hear from There ™ USER’S MANUAL Manual Version 3.0 for software version 3.0 and later (For 10 Base-T and 100 Base-T hardware versions)
Page 3: Customer Service
G ene ra l- vo n-Na ge l-St ras se 2 1 85354 Freising Germany Telephone: +49 81 61 42 467 Fax: +49 81 61 42 402 E- ma il a ddr es s: e uro pe- in fo@ telos-s ys te ms .co m INTRODUCTION & TABLE of CONTENTS...
Page 4 Updates The operation of the Telos Zephyr Xstream is determined largely by software. Periodic updates may become available. Contact us to determine if a newer release is more suitable to your needs. Trademarks Telos Systems, the Telos logo, Zephyr, ZephyrExpress, Zephyr Xstream, Liverwire and the phrase “The Best Way to Hear From There” are trademarks of TLS Corporation. All other trademarks are the property of their respective holders. Copyright Copyright © 1994, 2002, 2007, 2009 by TLS Corporation. Published by Telos Systems, who reserves the right to make improvements or changes in the products described in this manual, which may affect the product specifications, or to revise the manual without notice. All rights reserved. Notice All versions, claims of compatibility, trademarks, etc. of hardware and software products not made by Telos mentioned in this manual or accompanying material are informational only. Telos Systems makes no endorsement of any particular product for any purpose, nor claims any responsibility for operation or accuracy. Warranty This product is covered by a one year limited warranty, the full text of which is included in the Section 14 of this manual. Service You must contact Telos before returning any equipment for factory service. Telos Systems will issue a Return Authorization number, which must be written on the exterior of your shipping container. Please do not include cables or accessories unless specifically requested by the Technical Support Engineer at Telos. Be sure to adequately insure your shipment for its replacement value. Packages without proper authorization may be refused. US customers please contact Telos technical support at +1 (216) 241‐ 7225. All other customers should contact your local representative to arrange for service. We strongly recommend being at the unit when you call, so our Support Engineers can verify information about your unit and the conditions under which the problem occurs. If the unit must return to Telos, we will need your serial number, located on the rear panel. USER’S MANUAL INTRODUCTION & TABLE of CONTENTS...
Page 5 Test Lines To aid you in testing and demonstrating your Zephyr Telos Systems has the following test numbers available: • USA: +216 781.9310, +216 781.9311 (Layer III Dual transmit @ 32 KHz sample rate) • USA: +216 621.1826 (aacPlus mono, must call as an Xport call) • Germany: (49) 81 61 42 061 Dial this number twice (Layer III Dual transmit @ 32 KHz sample rate) • Germany: (49) 81 61 42 062 (Layer II Mono @ 32 KHz) USER’S MANUAL INTRODUCTION & TABLE of CONTENTS...
Page 6 Notices and Cautions CAUTION: THE INSTALLATION AND SERVICING INSTRUCTIONS IN THIS MANUAL ARE FOR USE BY QUALIFIED PERSONNEL ONLY. TO AVOID ELECTRIC SHOCK, DO NOT PERFORM ANY SERVICING OTHER THAN THAT CONTAINED IN THE OPERATING INSTRUCTIONS UNLESS YOU ARE QUALIFIED TO DO SO. REFER ALL SERVICING TO QUALIFIED PERSONNEL. WARNING: TO REDUCE THE RISK OF ELECTRICAL SHOCK, DO NOT EXPOSE THIS PRODUCT TO RAIN OR MOISTURE.
Page 7 USER’S MANUAL NOTE: Objectionable interference to TV or radio reception can occur if other devices are connected to this device without the use of shielded interconnect cables. FCC rules require the use of only shielded cables. CANADA WARNING: “This digital apparatus does not exceed the Class A limits for radio noise emissions set out in the Radio Interference Regulations of the Canadian Department of Communications.” “Le present appareil numerique n’emet pas de bruits radioelectriques depassant les limites ...
Page 8: Table Of Contents
QUICK START – Step by Step Guide to Getting Running .. 1 Connections Preview .. 2 Success, Step‐by‐Step: ... 4 Powering Up ... 6 Basic Configuration .. 7 Placing Your First Call (to Yourself) .. 10 Your Next Call, to Telos .. 11 Where to find answers to frequently asked questions: ... 13 INTRODUCTION ‐ Getting to Know the Zephyr Xstream ... 15 What is the Zephyr Xstream?... 15 Why the Zephyr Xstream? ... 16 Xstream vs. Xstream MX vs. Xstream MXP .. 16 GUIDED TOUR of the HARDWARE .. 19 ...
Page 9 AUDIO CODING REFERENCE .. 109 Introduction to Audio Coding Technology ... 109 Choosing the Best Coding Mode for Your Application ... 126 ISDN BASICS .. 133 Background .. 133 The ISDN Basic Rate Interface (BRI) .. 134 How to Order ISDN ... 138 Switched 56... 142 LIVEWIRE – IP Audio ... 143 Configuration of the Zephyr Xstream for Livewire .. 143 Livewire Audio Settings .. 149 The V.35/X.21 Interface Option .. 153 ...
Page 10 System Functions and Remote Control ... 245 12.1 Ancillary Data ... 245 12.2 Remote Control .. 247 DEEP TECH INFORMATION & TROUBLESHOOTING ... 261 13.1 Accessing the Hardware ... 262 13.2 Updating Firmware .. 265 13.3 Updating Xstreams with the 100 Base‐T hardware .. 265 13.4 Updating the original (10 Base‐T) Xstream ... 267 13.5 Troubleshooting .. 273 13.6 General Troubleshooting Techniques ... 273 SPECIFICATIONS & WARRANTY .. 293 ...
Page 11: Conventions Used In This Manual
This information is important to safe operation of the equipment and warns of possible hazards. For your safety you should read these. IMPORTANT TIP! This information is essential to getting the Zephyr Xstream to work, or prevent damage to it. To avoid headaches read these. HOT TIP! This information will probably come in handy at some point.
Page 12 Amaze your technical friends with your voluminous knowledge of the Zephyr Xstream, ISDN, and coding technology! Not necessary for the ability to install and use the Zephyr Xstream, but this information might come in handy if you ever need to troubleshoot the system.
Page 13: Quick Start - Step By Step Guide To Getting Running
QUICK START – Step by Step Guide to Getting Running Who can use this section? You have just opened the carton containing a cool new Telos codec. We know how you feel. You don’t want to sit around and read a manual. You want to plug‐in your new Zephyr Xstream and hear what it can do. You want to connect to a distant location and be amazed at how audio over ISDN connections or Ethernet is even better than ever. We know because we feel the same way when we get a new, expensive, piece of gear. As much as we understand your excitement, we ask you to please peruse the following to get the most out of your Zephyr Xstream. This manual is designed to fully instruct you on the capabilities of your Zephyr Xstream and to provide you with interesting and useful information on audio coding and digital telephony. (Unlike most other manuals, large sections don’t even require that you be sitting in front of the product.) We hope you will take the time to read the manual. We trust you will enjoy it and that it will provide you with new and useful information that you can use to impress others. If you need to put the unit in service today, or have experience with other, similar equipment, this section will allow you to get started quickly with your Zephyr Xstream. “Quick Start” Is for You If You... • Know how to dial a telephone call. • Are not intimidated by looking at the front panel of the Zephyr Xstream. ...
Page 14: Connections Preview
What you need before you start… There are two categories of items you need: First, the equipment to connect your Zephyr Xstream to the audio equipment and the ISDN circuit; second, some information that should have been provided to you by the person who provided your ISDN. Connections Preview ISDN Connections All of the connection cables have telephone‐type modular connectors and come packaged with your Zephyr Xstream. IMPORTANT TIP! An RJ-11 style 6-position plug can be inserted into the center of an RJ-45 style 8-position jack, if the Telco installed an RJ-45 style jack for your U Interface ISDN line (in the USA or Canada).
Page 15 Japanese ISDN do not have SPIDs! If your equipment configuration requires MSNs, they may be entered in the MSN/DN menu options in the TEL menu. MSNs are optional, would only be used if another ISDN device is sharing the ISDN circuit with the Zephyr Xstream. Here is a summary of what you need to know and a place to write it down. You may want to ...
Page 16: Success, Step-By-Step
If you have European, Japanese, or AT&T Point-to-Point ISDN, it is acceptable if you only have one number that applies to both lines. If you have another ISDN type and only one number, contact Telos Customer Support. Success, Step-by-Step: Connecting the Audio Inputs/Outputs Got your audio cables ready? Here is where to connect them: ...
Page 17 connectors labeled “OUTPUTS: RECEIVE FROM NETWORK.” If you are not certain about how to make this connection, refer to Section 3. Livewire If you are already familiar with Axia’s audio nodes, you might think of this new version of the Zephyr Xstream as an “original” Xstream codec with a 2x2 node on the front of it. The LAN port will be used for all Livewire audio connections in and out of the codec. As with all Livewire devices, the Livewire streams being delivered to the network require unique ID numbers. The two most important concepts to understand are: 1. The Livewire destinations deliver audio to the inputs to the Xstream codec. 2. The outputs of the Xstream codec (Hybrid Channels) will become the codec’s Livewire sources to your Axia Livewire network. The virtual 2x2 node, in combination with the codec, has some unique mono/stereo features: • If the stream being received by the codec is stereo ‐ both codec outputs (Livewire sources) will be stereo with both delivering identical stereo streams. • If the stream being received by the codec is dual mono (such as G.722), the decoded audio is split between the two Livewire replicated on both left and right channels. The second source has the B mono feed on both channels. Thus ‐ when you are using dual mono, you will use both codec outputs as independent Livewire sources. • Remember, Livewire streams are always stereo. In the case of a mono source as described above, the mono data is simply replicated on both left and right channels. • Dual/Mono TRANSMIT works only in Layer 3 Dual/Mono mode. In Layer 3 Dual/Mono transmit; the left channel of the first destination is used to create the “Channel A” mono feed to be sent over ISDN. The second destination is used for the “Channel B” mono transmit feed. The right channel of each of those streams is ignored. A Word about Livewire Codec Backfeeds ...
Page 18: Powering Up
Connecting to the ISDN Circuit Making the physical connection to the ISDN line varies with your NT1. You will use one of the following methods as appropriate for your locale: 1. Zephyr Xstream’s built‐in NT1 (i.e. USA & Canada) Using a standard RJ‐11 style telephone‐type modular cable, interconnect the smaller modular jack on the Zephyr Xstream and the jack installed by the telephone company. Be careful to insert the cable in the center of each jack, as it is the middle two wires that have the ISDN circuit. 2. External NT1 provided by the Telephone company (Europe & most locations worldwide) Using a large 8‐pin (RJ 45 style), telephone‐type modular cable with eight wires, interconnect the larger jack on the Zephyr Xstream and a jack on the NT1 that is labeled “Terminal” or "S/T." This cable must have eight wires and is provided with the Zephyr Xstream. You should not need to provide power to the NT1. The NT1 is normally powered from the ISDN line. Powering Up Time to turn on the Zephyr Xstream! • Connect the provided IEC power cable to the Zephyr Xstream. As you face the back, the connection is on the lower right. • Connect the other end of the IEC cable to your AC (mains) power outlet. The Zephyr Xstream’s power input is universal, accepting anything from 100 to 240 Volt AC 50/60Hz. • Flip the power switch, located just above the power cable connector, to the “on” position. (push in the side labeled “1”). • Your Zephyr Xstream will start up. After a few moments, the front panel LCD display screen will display the status of your Zephyr Xstream. Above the Send meters, the SYNC indicator should be illuminated. If you have a proper audio connection, and are feeding an audio signal to the Zephyr Xstream, ...
Page 19: Basic Configuration
Basic Configuration Your Zephyr should display a screen that looks like this: Press the <CODEC> button once. You should see a screen that looks like this: If it does not look like the above (i.e. some of the settings are different), your unit has been previously configured. If it was, and you want to proceed with the Quick Start, you must erase all of the previous settings. Before you go to the next step, check with the last person who used this Zephyr Xstream to be certain that s/he no longer needs these settings. To reset the Zephyr Xstream to the factory defaults: Press the <System> button (located beneath the LCD display) 5 times. You • will see a screen that looks like the following: Using the <6> button move the dark bar so that “Restore Factory Defaults” • is highlighted (white text on a black background). Press the <SEL> Button. You should see the following screen: Now press <6> then press <SEL> to erase all settings to the factory • defaults. We are now ready to configure your Zephyr Xstream. Follow these instructions exactly for the most reliable path to success! We will be using the <TEL> button. This will take you through the menus where the information about your ISDN circuit is stored. 1. You have two options depending on what type of ISDN you are using: USER’S MANUAL Section 1: QUICK RESULTS...
Page 20 USER’S MANUAL • If you have Euro‐ISDN (or another service that does not have SPIDs), press the < TEL > button (located beneath the LCD display) three (3) times and skip to step 5. • If you have SPIDs, press the < TEL > button two (2) times and continue. You should see a screen that looks like this: 2. Make sure the dark cursor bar is highlighting the line labeled “SPID 1”. Press the <SEL> button once and then enter the SPID of your first B channel, using the dial pad on the front right panel. (Note that there are no hyphens, punctuation, or blank spaces in SPIDs; they are one continuous number.) The <5> button can be used to back up (move the cursor to the left) to delete characters for editing (or to erase an existing entry). Then press the <SEL> button to store it. 3. Press the < 6 > button to highlight the line labeled “SPID 2 ”. Press the <SEL> button once and then enter the SPID of your second B channel using the dial pad on the front right panel, as above. Then press the <SEL> button to store it. 4. You now have two options: • If your ISDN does not require Directory Numbers (DN) press <TEL> one (1) time and skip to step 5. • If your ISDN requires Directory Numbers (DNs), and most users in the USA & Canada will, press the <6> button to move the dark cursor to the line labeled "DN/MSN 1". Press the <SEL> button once and then enter the DN (the 7 digit telephone number) of your first B channel using the dial pad on the front right panel. (Note that there are no hyphens, punctuation, or blank spaces in DNs or MSNs; they are one continuous number.) The <5> button can be used to back up (move the cursor to the left) and delete characters or erase and existing entry. Press the <SEL> button to store it. Press the <6> button to move the dark cursor to the line labeled "DN/MSN2". Press the <SEL> button once and then enter the DN or MSN of your second B channel using the dial pad on the front right ...
Page 21 5. Make sure the dark cursor bar is highlighting the line labeled “Telco”. Press the <SEL> button once. Press the <5> or <6> button repeatedly until the type of ISDN you have appears on the screen. Then press the <SEL> button to accept it. Even if the correct ISDN type is shown, press the <SEL> button to choose this item and then again to deselect it. You should now see the following screen: Press the <6> button and then <SEL> to confirm you wish to reboot. The Zephyr Xstream will reboot. After a few moments, you should see the following screen: In addition, you should observe the following: • The SYNC lamp will be glowing green near the top of your Zephyr Xstream’s front panel (above the Send meter). • If you are using the Zephyr Xstream’s internal NT1 (i.e. are a user in the USA or Canada), the little green LED (at the bottom left, of the back of the unit) should be illuminated solidly, not blinking or flickering. Note: if you are using an external (Telco provided) NT1 this LED will be flickering, this is normal. • Press the <TEL> button once. The two status words at the left of the LCD screen should both indicate “Ready” (you may need to wait as long as 30 seconds for the ISDN to initialize, so be patient). 6. If these conditions are met, you are ready to place your first call. If not, something is not right with your ISDN line, the way you have configured USER’S MANUAL Section 1: QUICK RESULTS...
Page 22: Placing Your First Call (To Yourself)
• Check all of your ISDN line connections to be certain they are firm and correct. Refer to the instructions above. Note: In the USA and Canada you should normally be using the smaller (RJ‐11 style) ISDN jack on the rear of the unit whereas users outside those countries will normally use the larger (RJ‐45 style) jack. • Your ISDN line may not be activated or may be configured incorrectly. Contact your ISDN line provider. US users should confirm that the line was installed exactly as specified in the ISDN ordering instructions found in the appendix of the manual. • If you are using Zephyr Xstream’s internal NT1, the green status LED (near the jack, on the rear panel) should be on solid. If it is not, you have a very basic line problem – see Section 13 (Deep Tech & Troubleshooting) for more advice and guidance. At this point, if the conditions in item 5 are still not met, contact Telos Customer Support for assistance. When you call, have the information on your ISDN circuit on hand. Placing Your First Call (to Yourself) Your first call is to be placed to yourself. You will be using your first ISDN "line" to connect to your second ISDN "line" (both "lines" are part of the same ISDN circuit). If this works, you know you are connected to your local telephone company office and your Zephyr is programmed correctly. (This procedure will not work if you only have ordered only one ISDN B channel from the Telco.) Follow these step‐by‐step instructions: 1. Press the <DIAL> button once. You will see a screen that looks like this: Since you are dialing from the first "line" (to your second "line"), use the keypad to enter the phone number of your second line (you do not need to press <SEL> before entering the number). If you need a code to access an outside line, be sure to enter it. USER’S MANUAL ...
Page 23: Your Next Call, To Telos
Note: Since this is a local call, you probably do not need a “1” or area code. 2. Press the <DIAL> button again. You should see a series of messages that say “Seize Pending; Out Ring; Connected.” If successful, the receive LOCK LEDs will light (it might take a few moments), and, if you are sending audio into your Zephyr, you will see activity on both the Send and Receive meters. If connected to your studio audio gear, you should be able to hear the audio feed from the “Output: Receive from Network” audio outputs, or your headphones. 3. Whether or not you have successfully connected, slowly press the <DROP> button repeatedly until both of the LINE status indicators have returned to “Ready”. This may take up to four presses. 4. If you have successfully connected, go on to the next section. If you have not, try again. Try dialing the other line number provided to you. You may wish to try with an area code or 1. If you fail after a few more attempts, contact Telos Customer Support or your local Telos Representative. Your Next Call, to Telos Located less than a mile from the Rock ‘n’ Roll Hall of Fame, there is always something interesting playing on the Telos ISDN test line. Let’s call Cleveland. If you prefer, you can call our test line in Europe. That number is listed under “Test Lines” on page iii at the front of this manual. Be certain to call a Layer‐3 line. Follow these step‐by‐step instructions: 1. Press the <DIAL> button once. Use the keypad to enter the first Telos test line number. Be certain to precede it with any number you may need to dial to access an outside line and/or to place a long distance or international call. The first number is +1 216.781.9310. Press the <DIAL> button again. 2. You should see a series of messages that say “Seize Pending; Outgoing Seize; Out Ring; Connected.” If successful, the green Lock LED over the left Send meter will be illuminated. You may not hear audio or see any meter action until you have connected the second line. If you do not connect, go to step 5, below. 3. Press the <DIAL> button once. Use the keypad to enter the second Telos ...
Page 24 You should see a series of messages that say “Seize Pending; Outgoing Seize; Out Ring; Connected.” If successful, the second green Lock LED over the right Send meter left of the LCD display will be illuminated. Unless the CD player at Telos has stopped, you should see activity on the receive meters and hear music. 4. Whether or not you have successfully connected, slowly press the <DROP> button repeatedly until both of the LINE status words have returned to “Ready”. This should take four presses. You want to be sure to disconnect, as this may be an expensive call. Alternately, you can select ‘Both’ lines from the drop menu: 5. If you have successfully connected, go on to the next section. If you have not, try again. If you do not succeed, here are a few things to try: • US users should try calling using another long‐distance carrier. To do so, insert the carrier’s seven‐digit access code. The number to dial starts with any prefix needed for you to get an outside line, followed by the long distance access code, followed by the Telos test line number. See Section 13 for additional tips. Let us say you don’t need a prefix (and most people don’t) and you want to use MCI. Dial: Line 1: 1010222 1 216.781.9310 Line 2: 1010222 1 216.781.9311 If this works, you should contact your phone company to determine how to resolve the problem with your primary long distance carrier. • Wait an hour and try again. The Telos test line may be busy. • Call Telos Customer Support for assistance. The terminal adapter in Zephyr Xstream has a built‐in ISDN analyzer that will provide plain language error messages in many circumstances. Have these messages ready when you call, as they will help speed the process of solving your problem. What’s Next? Now you are ready to call the world. Remember your promise to read the entire manual! If you ...
Page 25: Where To Find Answers To Frequently Asked Questions
Where to find answers to frequently asked questions: Question How do I order my ISDN line? What are the meanings of all the meters and indicators on the front panel? Should I use AAC, Layer III, Layer II, or G.722? How do I call a Zephyr Xport on a POTS line? How should the Zephyr Xstream be configured when the far end is not a Zephyr Xstream? How can I connect the Zephyr to a computer? How do configure the Xstream for Livewire? Can I call a regular Telephone? How do I contact Telos? Connector Pin‐outs? Specifications? W arranty? USER’S MANUAL Manual Section Appendices 7 (Ordering ISDN for ETS 300) & 8 (Ordering ISDN for USA & Canada) Section 3 (Guided tour to the hardware) Section 6 (Audio Coding Reference) & Section 10 (Applications Guide) Section 4.2 & Section 10.1 Section 10 (Applications Guide) ...
Page 26 USER’S MANUAL Section 1: QUICK RESULTS...
Page 27: Introduction - Getting To Know The Zephyr Xstream
A software upgrade in 2006 adds enhancements to IP such as bi‐directional dialing and AAC with error concealment. Units manufactured since 3 Livewire™ direct input/output to and from Axia audio networks. In short, the Zephyr Xstream is an advanced, economical, flexible, and easy to use audio codec. While we have added a number of advanced features, we’ve also maintained ease of use. Some highlights of the Zephyr Xstream family are: • MPEG 2 AAC (Advanced Audio Coding). This new standard for audio coding combines the techniques developed by researchers all over the world into a single algorithm. • Unique Dual Receive mode in MPEG Layer‐3, allows independent audio streams arriving from two distant codecs. • Unique POTS to ISDN technology communicates with a companion Xport in the field using aacPlus™ for the absolute best quality at the very low bitrates required. • Auto Receive mode searches and determines the correct decoder settings for the incoming audio stream. • Advanced technology – lower heat generation and improved reliability – Unit has no cooling fan for silent operation in your studio. • V.35 option allows connection to serial synchronous data equipment for use with services such as dedicated lines, Switched 56, or Satellite services. Section 2: INTRODUCTION – Getting to Know the Zephyr Xstream 15 USER’S MANUAL quarter 2006 also have hardware support for ...
Page 28: Why The Zephyr Xstream
• Supports the MPEG AAC, Layer‐2, Layer‐3 and G.722 coding standards. • MPEG AAC‐LD offers Layer‐3 fidelity with 75% less delay. • AES/EBU inputs and outputs (standard) • Analog line level inputs and outputs (Switchable between “Consumer” and “Professional” Levels) • Short learning curve. Zephyr users will be up and running in minutes. • High reliability design. No fan or battery to fail. Rugged chassis design. Zephyr Xstream MX and MXP These are typically found in remote facilities. These two units are identical with the exception that the MX is rack mountable whereas the MXP has a portable chassis. Highlights of the Zephyr Xstream MX & MXP are: • Supports the MPEG2 AAC, Layer‐2, Layer‐3 and G.722 coding standards. • Exclusive MPEG4 AAC‐LD offer Layer‐3 fidelity with 75% less delay. • Short learning curve. Zephyr users will be up and running in minutes. Section 2: INTRODUCTION – Getting to Know the Zephyr Xstream 16 USER’S MANUAL...
Page 29 USER’S MANUAL • 4 input stereo mixer. Each input gain selectable from mic. to line levels and is pan‐able left/right. Mixer directly feeds the audio coder section. • Exclusive, DSP‐based, audio processing by Omnia for AGC/Limiting • Selectable phantom power feeds inputs 1 & 2 • Two local mixes allow any combination of send and receive audio for local headphone monitoring or a Public Address feed. • Local Mix one appears on the front panel on a stereo ¼” connector. • Local Mix two appears on three front panel ¼” stereo connectors, each with a separate volume control. This mix is also available on a pair of balanced XLR outputs. • Rugged chassis design incorporates rubberized shock protection. Section 2: INTRODUCTION – Getting to Know the Zephyr Xstream 17...
Page 30 USER’S MANUAL Section 2: INTRODUCTION – Getting to Know the Zephyr Xstream 18...
Page 31: Guided Tour Of The Hardware
GUIDED TOUR of the HARDWARE The Zephyr Xstream, Xstream MX, and Xstream MXP are all quite similar in their design and operation. The menus are identical between all three versions with the exception of the AUDIO menu. The front and rear panels of the Xstream MX and Xstream MXP are identical. The only difference between these two models is the fact that the Xstream MX is rack‐mountable, whereas the Xstream MXP has a rugged portable case. The Xstream has different front and rear panels from the other models. Therefore, we will cover the hardware of the Xstream in Sections 3.1 (front) and 3.2 (rear) and the Xstream MX/MXP in Sections 3.3 (front) and 3.4 (rear). Zephyr Xstream Front Panel Input & Output Meters Sync Light Dual Lock Lights Operator has ready access to system operating information Headphone Jack & Volume Control USER’S MANUAL...
Page 32 USER’S MANUAL Headphone Jack & Volume Control The front panel has a headphone jack near the left side. The adjacent knob controls the level. This jack duplicates the receive signal on the main outputs on the rear panel. \ Meters and indicators The following LEDs allow you to monitor the system status and audio levels: Sync LED The green Sync LED indicates, “all is well” with the system. It indicates that certain clock signals are present and that the internal PLL is within its normal range. This LED should normally be illuminated at all times if the internal ISDN interface is functioning and selected as your Interface in the TEL Menu. When the V.35/X.21 interface is present (and selected) the Sync LED will be illuminated only if a valid external clock signal at the correct frequency for the Bit Rate setting (in the CODEC menu) is present on one of the two V.35/X.21 ports. When the Ethernet interface is selected, the Sync LED will blink while the Xstream’s clock works to sync with the Xstream on the other side of the connection. Due to network conditions, it is common for this light to be solidly on, then start blinking again as the clock readjusts. Lock 1 & 2 LEDs The green Lock 1 and Lock 2 LEDs represent the codec decoder status. These will illuminate when a valid bitstream is received as follows: If a dual channel receive mode is used (i.e. L3 Dual/Mono, or G.722) the Lock 1 LED represents that the decoder is locked (or “framed”) to the incoming coded audio on “Line 1” while the Lock 2 LED represents a decoder lock on an incoming audio stream on “Line 2”. In the case of a stereo, or mono‐128, receive mode, both lock LEDs will illuminate only once a compatible pair of bit‐streams are present on both “Lines”. Send Bargraph The Send Bargraph LEDs represent the level of audio into the coder section. This level is ...
Page 33 HOT TIP! Note that the CODEC, TEL, & SYSTEM menus (and Navigation and Menu Selection Keys) are identical among all models of the Zephyr Xstream. Only the AUDIO menu varies between the Xstream (non-mixer) and Xstream MX/MXP (mixer) versions. Navigation and Menu Selection Keys These keys are used to move through the menu options on a given page of menu ...
Page 34 Numeric Keypad The operation of this is straightforward. A given button corresponds to the number printed on it. It can also be used for text entry of the letters associated with it. Additional characters are available on the <*> key (i.e. * ( ) [ ] < > <space>) and the <#> key (i.e. # . , ; : ! ? <space>) and underscore is available on the <1> key. To enter text characters, press the key with the appropriate letter printed near it. Keep tapping it to cycle through the possible choices. For example, tapping the 2‐ key in a text field will first enter A. Repeated taps will change that to a B, then C, then the lower‐case letters, then the number 2. Enter a space by pressing 1 three times. Punctuation characters appear on the * and # keys. The period “.” is on the # key. For example: Press the number 2 once .. See the character A ... press it yet again .. the cycle starts again from A To move to the next character in a field, tap any other key or press the < > press the AUTO, DIAL, & DROP Keys These keys will be explained in detail in Section 4.2. The <AUTO> key allows access to Zephyr Xstream’s 100 stored Dial Setups (#00 to #99). These ...
Page 35 IMPORTANT TIP! The Number field (in the Dial menu) is different from other entry fields in the Zephyr Xstream. It is NOT necessary to press <SEL> before entering the number to be dialed. While we like consistency, this field is used so often, it made sense to make it easier by doing this.
Page 36: Zephyr Xstream Rear Panel (Non-Mixer Version)
USER’S MANUAL Zephyr Xstream Rear Panel (non-mixer version) Section 3: GUIDED TOUR of the HARDWARE 24...
Page 37 ISDN Interface IMPORTANT! As with any piece of modern electronic gear, it is advisable that precautions be taken to prevent damage caused by power surges. Standard telephone line surge protectors can be used to offer some degree of protection at the U interface jack. Special protectors are available for the S interface.
Page 38 HOT TIP! Normally, the Telephone Company should supply an RJ-11 style jack for the U interface. This is the same 4 or 6-pin plug used for analog lines. However, sometimes you may find an 8-pin RJ-45 style jack–and sometimes you may even be charged extra for it! If you have this situation, an RJ-11 plug can be inserted into the center of an RJ-45 jack to make things work.
Page 39 IMPORTANT! Do Not enable PS2 power if you plan to connect the S interface of your Xstream to a Zephyr Classic or ZephyrExpress. Doing so could cause damage to one of the units! ISDN TIP! If you are connecting to ISDN using the S (4-wire) Interface you can ignore the status of the NT1 U Sync LED.
Page 40 ZEPHYR CLASSIC VS XSTREAM COMPARISON! NOTE: The RS-232 connectors on the newer Telos gear, such as the Zephyr Xstream, are wired to look like a modem (i.e. DCE), rather than a computer. Therefore, a standard “straight through” cable can be used to connect the Xstream to your computer.
Page 41 Our thanks going out to Mike Schweizer, for contributing this information. USER’S MANUAL CABLE PIN OUT 9-PIN Direction CABLE PIN OUT 25-PIN Direction Section 3: GUIDED TOUR of the HARDWARE 29 Telos Xstream Not used Not used Not present Telos Xstream Not used Not used...
Page 42 They are not assigned to any router on the Internet. Of course, you should make sure they are not being used by another computer on your LAN. Note that these IP addresses cannot be used to update your unit directly from Telos (since they work for local access only).
Page 43 Parallel Control Port The parallel control port has 8 inputs and 8 outputs that can be used in a variety of ways. For instance, in AAC, AAC‐LD, Layer 3 and certain Layer 2 modes, these contact closures can communicate between two Zephyr Xstreams. By grounding an input on the local Xstream, you would therefore cause current to flow into the corresponding output, at the far end unit. These can also be used for local control and to monitor local status see Section 11 has additional information on the use of this port. ZEPHYR CLASSIC VS. XSTREAM COMPARISON! The original Zephyr has only four inputs and outputs numbered 0-3 while the Xstream has 8 inputs 0-7. The four inputs and outputs of the Zephyr will communicate with the first four inputs and outputs of the Xstream (i.e.
Page 44 Parallel Outputs 0‐7 Outputs are open collector to ground and can sink up to 125 mA of current each (actually, any output can sink up to 400 mA, however the maximum combined current should be kept below 1000 mA). IMPORTANT! Outputs on the Zephyr Xstream motherboards prior to Rev G (1401-0000--007) cannot be used with external pull-ups higher than 5 volts. If this ability is required contact Telos Systems Customer Support. These include a 10k‐ohm 5 volt pull‐up resistor to allow direct interface with 5 volt logic inputs on other equipment. If your other equipment uses a voltage greater than 5 volts, you will need an external pull up resistor, unless it is built‐in to that equipment's inputs. An appropriate value is in the range if 2.2KΩ to 10KΩ. Operation with logic inputs requiring a voltage less than 5 volts ...
Page 45 (30 Volts max) PIN 25 DEEP TECH NOTE! The Zephyr Xstream’s “universal” logic input circuit (motherboard rev G and later) can be used with switch or relay closures, voltage levels up to 24 Vdc, or logic outputs – either “totem-pole” or open-collector.
Page 46 XLR PIN The analog audio inputs have the following characteristics: • Active balanced • Line level: A menu selection Level In, in the AUDIO menu, allows selection for Consumer or Pro levels. The nominal input level is jumper settable on the audio board (See Section 13.1) to allow the following nominal levels: JUMPER SETTING (J10) *Pins 1/2 and 5/6 Pins 2/3 and 4/5 Factory default setting • Headroom: 18dB above nominal • Bridging ≥ 10 K ohm impedance An analog soft clipper is included to allow maximum dynamic range, while still protecting the user from over loading the A/D converter if input signal exceed these values. The inputs are designed to be sourced from balanced, line level, signals. Older equipment, with a transformer output stage, may need a terminating resistor (usually 600 ohms) across pins 2 and 3, consult the manual for your equipment for how to use it with high impedance inputs. Unbalanced sources may be used by connecting pins 3 to the source ground while the signal high is connected to pin 2. We recommend leaving pin 1 (ground) unconnected, as this arrangement will prevent the possibility of ground loops. HOT TIP! Xstreams with Stereo Audio boards prior to Rev G (1401-0009-007) can handle unbalanced signals at -11dBu or -15dBu only (see jumper settings, above).
Page 47 CURIOSITY NOTE! The Telos Zephyr Xstream uses the more common pin-outs for three pin XLR inputs & outputs. You can easily remember the correct signals when wiring connectors using the phrase “George Washington Bridge.” Pin 1 = G = Ground, Pin 2 = W = “+” = White (typical color in mic cable, if there is no white there will be a red conductor), and Pin 3 = B = “-”...
Page 48 “clock slip” due to different clock rates within a digital facility. Typically a “house clock” reference would be fed to the SYNC IN jack of the Zephyr Xstream, as well as to other equipment, to prevent drop outs due to buffer “overflow” or “underflow” conditions.
Page 49 AC (Mains) power The AC receptacle connects mains power to the unit with a standard IEC (International Electrotechnical Committee) power cord and provides an on/off switch. The power supply has a “universal” AC input, accepting a range from 100 to 240 VAC, 50‐60 Hz. A fuse is located inside on the power supply circuit board. IMPORTANT! As with any piece of modern electronic gear, it is advisable that precautions be taken to prevent damage caused by power surges. Standard line surge protectors can be used to offer some degree of protection. It is the user’s responsibility to ensure protection adequate for their conditions is provided.
Page 50 Zephyr Xstream MX/MXP Front Panel 4 Input Stereo Mixer Feeds codec section directly. Each Mic../Line Input panable Left/Right. Inputs 1&2 support 48volt phantom pow er. All controls, including flexible limiter/processor accessed through Audio Menu. Local Monitor 1 Local Monitor 2 Jack &...
Page 51 Local Monitor Mix 2 Volume Controls Local Mix 2 is available on three rear‐panel headphone jacks (1/4” stereo jacks) as well as, by default, a pair of balanced XLR jacks ‐ we’ll cover these jacks shortly, in Section 3.4. Each of these headphone jacks has an associated front‐panel volume control. Just as with Local Monitor Mix 1, an adjacent button allows you to select if you wish to monitor Snd (send) audio, Rcv (receive) audio or a mix of both. Repeatedly pressing this button cycles between the options. The seventh AUDIO menu allows you to set up the Send/Receive and Left/Right balance for local monitor mix 2. Main Mixer Controls These for knobs control the Zephyr Xstream MX/MXP’s input levels for the main mixer section. A button adjacent to each input allows assignment to the A (Left), B (Right), or both, mixer busses. • The nominal input gain and AGC/Limiter options for each mixer input can be set on the second through the fifth AUDIO menus (for inputs 1‐4, respectively). • The Main Mixer's output directly feeds the codec section. • The factory defaults for inputs 1 and 2 is for microphone inputs levels and assigned to both A and B audio channels. • The factory defaults for input 3 is "professional" input levels assigned to the A audio channel. • The factory defaults for input 3 is "professional" input levels assigned to B audio channel. USER’S MANUAL Section 3: GUIDED TOUR of the HARDWARE 39...
Page 52 HOT TIP! The nominal inputs levels for each input can be adjusted using the Ch "x" Gain option in the 2nd through 3rd AUDIO menus. You can also enable phantom power for inputs 1 & 2 on AUDIO menu 8. Note that this controls phantom power for both input 1 &...
Page 53 When the V.35/X.21 interface is present (and selected) the Sync LED will be illuminated only if a valid external clock signal at the correct frequency for the Bit Rate setting (in the CODEC menu) is present on one of the two V.35/X.21 ports. When the Ethernet interface is selected, the Sync LED will blink while the Xstream’s clock works to sync with the Xstream on the other side of the connection. Due to network conditions, it is common for this light to be solidly on, and then start blinking again as the clock readjusts. Lock 1 & 2 LEDs The green Lock 1 and Lock 2 LEDs represent the codec decoder status. When a dual channel receive mode is used (i.e. L3 Dual‐mono or G.722) the Lock 1 LED represents that the decoder is locked (or “framed”) to the incoming coded audio on “Line 1” while the Lock 2 LED represents a decoder lock on an incoming audio stream on “Line 2”. In the case of a receive mode requiring both B channels (most stereo mode as well as mono 128) both lock LEDs will illuminate only once a compatible pair of bit‐streams are present on both “Lines”. Send Bargraph The Send bargraph LEDs represent the level of audio into the coder section. This level is affected by several factors: The output level of the device connected to the Zephyr Xstream; the setting of the Ch "x" Gain option in the AUDIO menus; the mixer knob setting. As with most digital equipment, these meters are calibrated in dBfs (decibels below full scale). In other words, the top LED represents the clip point and the numbers are calibrated in dB below full scale. Therefore, it is important that your levels be set to where the top LED is never lit. There is a transition of LED color at ‐10 dBfs. Since the meters are peak indicating, it will generally be safe to run things into the red, depending on the desired headroom and the dynamics of the material. When setting levels with a sine wave (tone), a level near the red green boundary should be used. HOT TIP! The individual level for the audio of each of the 4 mixer inputs (post fader) can be monitored on the first AUDIO menu.
Page 54 HOT TIP! Note that the CODEC, TEL, & SYSTEM menus (and Navigation and Menu Selection Keys) are identical among all models of the Zephyr Xstream. Only the AUDIO menu varies between the Xstream (non-mixer) and Xstream MX/MXP (mixer) versions. Navigation and Menu Selection Keys These keys are used to move through the menu options on a given page of menu ...
Page 55 then the lower‐case letters, then the number 2. Enter a space by pressing 1 three times. Punctuation characters appear on the * and # keys. The period “.” is one the # key. For example: Press the number 2 once .. See the character A ... press it yet again .. the cycle starts again from A To move to the next character in a field, tap any other key or press the <6> button. To back up (delete), press the <5> key. AUTO, DIAL, & DROP Keys These keys will be covered in greater detail in Section 4.2 (Basic Installation & Operation). AUTO The <AUTO> key allows access to Zephyr Xstream’s 100 stored Dial Setups (#00 to #99). These can be accessed by pressing <AUTO>, entering two digits, and then pressing <AUTO> again. It can also be used to access the 30 Location Setups (#100 to 129) can be accessed to configure the ISDN settings for frequently visited locations. Press and hold the <6> key to jump to the Location Setups. Or, press <AUTO> enter 3 digits and press <AUTO> again. DIAL The <DIAL> key is used to manually dial a number. Press, <DIAL>, enter the phone number and press <DIAL>. To redial the last number dialed simply press <DIAL> twice. DROP The <DROP> key is also straightforward in its most basic form. Pressing <DROP> twice will drop ...
Page 56: Zephyr Xstream Mx/Mxp Rear Panel (Mixer Versions)
USER’S MANUAL Zephyr Xstream MX/MXP Rear Panel (Mixer versions) Section 3: GUIDED TOUR of the HARDWARE 44...
Page 57 ISDN Interface IMPORTANT! As with any piece of modern electronic gear, it is advisable that precautions be taken to prevent damage caused by power surges. Standard telephone line surge protectors can be used to offer some degree of protection at the U interface. Special protectors are available for the S interface.
Page 58 HOT TIP! Normally, the Telephone Company should supply an RJ-11 style jack for the U interface. This is the same 4 or 6-pin plug used for analog lines. However, sometimes you may find an 8-pin RJ-45 style jack–and sometimes you may even be charged extra for it! If you have this situation, an RJ-11 plug can be inserted into the center of an RJ-45 jack to make things work.
Page 59 ZEPHYR VS XSTREAM COMPARISON! NOTE: The RS-232 connectors on the newer Telos gear, such as the Zephyr Xstream, are wired to look like a modem (i.e. DCE), rather than a computer. Therefore, a standard “straight through” cable can be used to connect the Xstream to your computer.
Page 60 "E" and later Using a 9 pin female D-Sub connector, this is an RS-232 serial port using the standard PC- style format, configured as if it were a modem. Zephyr Xstream ignores any incoming A computer plugged into this port can control the system, and can access a number of special diagnostic modes, as described in Section 12 (System Functions & Remote Control). It is also ...
Page 61 Our thanks going out to Mike Schweizer, for contributing this information. USER’S MANUAL CABLE PIN OUT 9-PIN Direction CABLE PIN OUT 25-PIN Direction Section 3: GUIDED TOUR of the HARDWARE 49 Telos Xstream Not used Not used Not present Telos Xstream Not used Not used...
Page 62 They are not assigned to any router on the Internet. Of course you should make sure they are not being used by another computer on your LAN. Note that these IP addresses cannot be used to update your unit directly from Telos (since they work for local access only).
Page 63 Parallel Control Port The parallel control port has 8 inputs and 8 outputs that can be used in a variety of ways. For instance, in AAC, AAC‐LD, Layer 3 and certain Layer 2 modes, these contact closures can communicate between two Zephyr Xstreams. By grounding an input on the local Xstream, you would therefore cause current to flow into the corresponding output at the far end unit. These can also be used for local control and to monitor local status. ZEPHYR CLASSIC VS. XSTREAM COMPARISON! The original Zephyr has only four inputs and outputs numbered 0-3 while the Xstream has 8 inputs 0-7. The four inputs and outputs of the Zephyr will communicate with the first four inputs and outputs of the Xstream (i.e.
Page 64 Parallel Outputs 0‐7 Outputs are open collector to ground and can sink up to 125 mA of current each (actually, any output can sink up to 400 mA, however the maximum combined current should be kept below 1000 mA). IMPORTANT! Outputs on Zephyr Xstream motherboards prior to Rev G (1401-0000--007) cannot be used with external pull ups higher than 5 volts. If this ability is required, please contact Telos customer support. These will include a 10k‐ohm 5 volt pull‐up resistor to allow direct interface with 5 volt logic inputs on other equipment. If your other equipment uses a voltage greater than 5 volts, you will need and external pull up resistor, unless it is built‐in to that equipment's inputs. An appropriate value is in the range if 2.2K voltage less than 5 volts is not possible. ...
Page 65 INPUT (30 Volts max) PIN 25 DEEP TECH NOTE! The Zephyr Xstream’s “universal” logic input circuit can be used with switch or relay closures, voltage levels up to 24 Vdc, or logic outputs – either “totem-pole” or open- collector. The inputs can be transmitted to the far end Zephyr Xstream when using the Layer‐3 or AAC ...
Page 66 Main Mixer Inputs (Send to Network) These are Combo Connectors with XLR female and ¼” Tip‐Ring‐Sleeve (TRS) female inputs. The pin out is as follows: XLR PIN FUNCTION The analog audio inputs have the following characteristics: • Active balanced • Mic. or Line level: The system allows selection from 2 microphone settings and two line level settings, as set using the Ch X Gain selection in the 2nd through 5th AUDIO menus. The mix level for each range is adjustable, of course, and is also affected by the built‐in AGC/Limiter (selected in the Ch "x" Proc menu selection on the second through the fifth AUDIO menus). The values given above are the nominal levels for a sine wave when the input adjustment is at the 3‐o'clock position and the AGC/Limiter is off. The following are the nominal inputs levels for each setting (AGC/Limiter off, fader set to mid position). 15 dB of headroom is provided. • With certain high output microphones, a 10 dB pad may be necessary to prevent overload. These are commonly available: For instance the Whirlwind Pad IMP‐10. MENU SETTING (AUDIO menus 2-5) MIC +44 MIC +30 Line Con Line Pro USER’S MANUAL TRS PIN Ground...
Page 67 See section 10.3 for details on dealing with delay and mix-minus. CURIOSITY NOTE! The Telos Zephyr Xstream the more common pin-outs for three pin XLR inputs & outputs. You can easily remember the correct signals when wiring connectors using the phrase “George Washington Bridge.”...
Page 68 Local Monitor Mix 2 can be set up to monitor Snd (send) Audio, Rcv (receive) Audio or a combination of both. The audio to be monitored is controlled by the button adjacent to the headphone 4 knob on the front panel. The ratio of send to receive audio and other parameters are set up in the seventh AUDIO menu. Headphones 2 – 4 have individual front panel volume controls. The A and B line outputs are at +4 dBu nominal level. These are standard ¼” stereo jacks. They are designed to handle headphones with impedances down to 10 ohms. The output level is approx 3 watts into 20 ohms and will vary depending on the headphone impedance. XLR Monitor Mix Outputs By default these XLR outputs are fed from Local Monitor Mix 2. On units with rev F or later MIC Audio Boards (part # 1401‐00008‐006 or later) the unit can be configured to be fed from Local Monitor Mix 1. These are XLR male sockets with the following pin outs. The analog audio outputs have the following characteristics: • Active differential • Output level; Line Level output at +4 dBu nominal. The Level Out option in the eighth AUDIO menu allows selection for Consumer (‐10 dBu) or Professional (+ 4 dBu) output levels for these outputs. 18 dB headroom is provided (+22 dBu maximum output level). • Impedance: < 33 ohm x 2 If a single‐ended (unbalanced) output is required, connect between ground and either of the output pins. Do not ground the unused pin. Use the same pin on both outputs to maintain phase. AC (Mains) power The AC receptacle connects mains power to the unit with a standard IEC (International Electrotechnical Commission) power cord and provides an on/off switch. The power supply has a “universal” AC input, accepting a range from 100 to 240 VAC, 50‐60 Hz. A fuse is located inside on the power supply circuit board. USER’S MANUAL FUNCTION Ground Audio + Audio - Section 3: GUIDED TOUR of the HARDWARE 56...
Page 69 IMPORTANT! As with any piece of modern electronic gear, it is advisable that precautions be taken to prevent damage caused by power surges. Standard line surge protectors can be used to offer some degree of protection. It is the user’s responsibility to ensure protection adequate for their conditions is provided.
Page 70 USER’S MANUAL Section 3: GUIDED TOUR of the HARDWARE 58...
Page 71: Basic Operation & Installation
USER’S MANUAL BASIC OPERATION & INSTALLATION This section is intended only as an introduction. For many ISDN users, it will provide all the information needed to get started with the unit. However, you may also need to review Section 5 (Guided Tour of the Menus) & Section 11 (Detailed Menu Reference), particularly if you are using remote control or one of the other advanced features. If you are using V.35 or Ethernet streaming mode see Section 10 and Appendix 6, respectively. Section 7 also has a number of tips about SPIDs and ISDN protocol types. Section 1 has step‐by‐ step instructions for quick start up and initial testing. Essentials Your Zephyr Xstream should ideally be placed in a location convenient to the operator. As the unit has no fan or other noise‐producing elements, you may wish to locate it in the studio to which it is connected. Large facilities may wish to locate a bank of Zephyr Xstreams in the main equipment room. In this case a web browser can be used to control the Zephyr Xstream remotely using HTML. Or, you may wish to consider using The Software Authority’s Zephyr Remote software package (see http://www.softwareauthority.com/zephyrremote.htm for additional information). Environmental Considerations Heating, Ventilating, & Cooling Requirements For optimal reliability, it is necessary to avoid exceeding the maximum allowable ambient air temperatures surrounding the Zephyr Xstream. Ambient air temperature should be between 0 to 40 degrees Celsius (32 to 104 degrees Fahrenheit). Relative humidity may be between 0 to 98% (non‐condensing). AC (mains) Requirements The system is designed to work from 100 to 240 volts AC, 50‐60 Hz. As with any microprocessor‐ based equipment, it is desirable that the Zephyr Xstream be fed from a reliable source of clean power. A UPS (uninterruptible power supply) with noise suppression would be helpful, particularly if local power is unreliable. AC surge protection is also desirable. Our first choice would be a top‐of‐the‐line unit containing a ferro‐resonant transformer as well as a UPS. You will want to avoid "bargain basement" ...
Page 72 It is essential that the third “grounding” pin not be defeated on the power cord and that the power cord be connected to a properly grounded receptacle. If a grounded receptacle is not available, a qualified electrical contractor, familiar with the regulations in your area, should be contracted to provide one. IMPORTANT! As with any piece of modern electronic gear, it is advisable that precautions be taken to prevent damage caused by power surges. Standard line surge protectors can be used to offer some degree of protection. It is the user’s responsibility to ensure protection adequate for their conditions is provided (see above).
Page 73: Introduction To Zephyr Xstream's Menu System
Another source would be Panamax. Their model “Max ISDN” Allpath is suitable for the ISDN S or U interfaces (also suitable for T1). Another source is Allcom model "552‐U", suitable for the ISDN S and U interfaces. See: http://www.allcomtlc.com Another possible source for an appropriate unit is Polyphaser. See http://www.polyphaser.com/ Users have reported good results with their multi‐conductor twisted pair models. (a unit with 65 kHz bandwidth should be correct) They also have a number of good papers on lightning protection on their site. A source for protection on Switched‐56 or DDS circuits (model 22‐64) protectors is Telebyte corp. See http://www.telebyteusa.com . Many other sources are available. Search "surge+ISDN" for numerous sources. Introduction to Zephyr Xstream’s Menu System The Zephyr Xstream has seven menus. The four primary menus are AUDIO, CODEC, TEL and SYSTEM. Repeated presses of the associated menu button, located immediately below the LCD screen, access these menus. The following information is meant as a quick introduction only. Section 11 (Detailed Menu Reference) covers the menu selections for the primary menus in detail. The Default Display The default display will appear after the system has been left idle for 60 seconds. It gives summary information on system status. At the bottom, in inverse text is the status of each ISDN "line" or V.35 port. The Primary Menus; AUDIO, CODEC, TEL & SYSTEM ...
Page 74 IMPORTANT TIP! Like many devices using telephone keypads, the Zephyr Xstream uses multiple presses of the various buttons of the keypad to enter all of the letters of the alphabet, as well as the digits 0 through 9. We will cover this in detail below.
Page 75 Let’s go through a couple of quick examples to see the Zephyr Xstream menus in action. Example 1: Selecting a setting from a list (i.e. the bit rate): 1. Press the <CODEC> button twice. You should see a screen that looks like the following: 2. Now press the <6> button twice so the menu item “bitrate” is highlighted (i.e. is shown with white text on a black background) as shown below: 3. Press the <SEL> button once. Note that the menu item moves to the top of the screen and information about “Bitrate” is displayed. 4. Press the <6> button repeatedly to view the possible settings for this menu item (i.e. 56 or 64). IMPORTANT TIP! You can press the <CODEC> button at any time to “escape” without saving the changes you have made.
Page 76 Example 2: Entering a numeric entry (i.e. entering a SPID): 1. Press the <TEL> button twice. You should see a screen that looks like the following: 2. Verify that the menu item “SPID 1” is highlighted (i.e. is shown with white text on a black background). If not, use the <5> button to highlight this menu selection. 3. Press the <SEL> button once. Note that the menu item moves to the top of the screen and information about “SPID 1” is displayed. 4. Press the <5> button repeatedly to erase any existing information. 5. Type in the new SPID number using the Zephyr Xstream’s keypad. IMPORTANT TIP! You can press the <TEL> button at any time to “escape” without saving the changes you have made.
Page 77 the 2‐key in a text field will first enter A. Repeated taps will change that to a B, then C, then the lower‐case letters, then the number 2. Enter a space by pressing 1 three times. Punctuation characters appear on the * and # keys. For example: Press the number 2 once .. See the character A ... press it yet again .. the cycle starts again from A To move to the next character in a field, tap any other key or press the <6> button. To back up, press the <5> key. Section 4: BASIC OPERATION & INSTALLATION 65 USER’S MANUAL ... press it again .. See the character B ... press it again .. See the character C ... press it again ...
Page 78 Special Menus; DIAL, AUTO & DROP The DIAL menu The DIAL menu appears after the Dial button has been pressed and looks like the following: If you wish to dial a data call to another ISDN codec (Mode = Zephyr) you can simply enter the desired number and press <DIAL> again. A second call can be placed on "Line 2" by pressing <DIAL> a third time. To drop all calls in progress, simply press <DROP> four times: This sequence is illustrated here. Section 4: BASIC OPERATION & INSTALLATION 66 USER’S MANUAL Ready Ready DIAL Phone # DIAL Connected: 0:00 Connected: 0:00 DROP Four Times...
Page 79 IMPORTANT TIP! The Number field in the DIAL menu is different from other entry fields in the Zephyr Xstream. It is NOT necessary to press <SEL> before entering the number to be dialed. While we like consistency, this field is used so often, it made sense to make it easier by doing this.
Page 80 V.35 Split, or V.35 single, the system will change the interface to ISDN so that the number(s) may be dialed. ZEPHYR CLASSIC VS XSTREAM COMPARISON The Zephyr Xstream’s Dial Setups offer some important differences from the Zephyr’s Auto Dial Setups: 1. Inclusion of codec information is optional with the Xstream, whereas it is mandatory with the Zephyr.
Page 81 HOT TIP! You can use dial setups as a way to conveniently change your codec settings. Possibly to accommodate incoming calls from differing locations. Simple leave the Number 1 & Number 2 field blank, and set Include to Codec. As always, be sure to set the CODEC menu items to the proper settings before creating your setup.
Page 82 The illustration below shows several sample Dial Setups. Note that the notation “C” in the right column indicates that a particular setup includes CODEC menu settings. Creating Dial Setups – Step by step 1. Configure the Zephyr Xstream’s CODEC menu as it will be used when connected to this location. This step is optional, you can skip this step if you are creating a Dial Setup that does not include codec settings HOT TIP! You will probably wish to include the CODEC settings in your Dial Setup. This allows you to dial a variety of locations, each with differing codec requirements, without the need for manual configuration.
Page 83 HOT TIP! You can use a Dial Setup to easily configure your Zephyr Xstream for inbound calls. To do so, just leave the Number 1 & Number 2 fields empty and set Include to "Number & Codec". When this dial setup is activated, the Xstream will change CODEC Menu settings to the settings present when this dial setup was created.
Page 84 Editing Dial Setups 1. If this setup includes codec information, configure the Zephyr Xstream’s CODEC menu as it will be used when connected to this location. If the existing codec information included with this setup is correct, simply activate the setup to recall those settings now (then drop the call (s) by pressing drop 4 times). If you are editing a Dial Setup that does not include codec settings you can skip this step. 2. Press the <AUTO> key. Choose the Dial Setup to be edited. To choose a Dial Setup you have two options: • Use the <5> and <6> to scroll through the list of 100 Dial Setups. • Or: Use the Keypad to enter the 2‐digit number (00 to 99) for the desired setup. That setup will be displayed and will be highlighted. 3. Once the desired setup is highlighted, press the <SEL> button. The name of the setup will be displayed. 4. Press <5> and then <SEL> to edit the setup. 5. Edit the name, number 1 or 2, or call type 1 or 2 (see the beginning of this section for an example on how to select and enter information). 6. You can choose whether to Include Codec settings or not by choosing this menu selection. The choices are Number (no codec information included with this dial setup) or Number & Codec. If Number & Codec is selected, the system will include the current settings in the CODEC Menu with this setup. Each time this setup is activated, the system will reset the CODEC Menu items to these setting before dialing the number(s) for this setup. Section 4: BASIC OPERATION & INSTALLATION 72 USER’S MANUAL ...
Page 85 7. Move the black cursor to the word Save and then press <SEL> to complete the editing process. Location Setups Location setups are a useful feature that allows you to pre‐configure information about the ISDN line (TEL menu information) and the mixer setup/assignment information (AUDIO menu and MX/MXP front panel information) for frequently visited locations. Each dial setup consists of several pieces of information: • A Setup number – this three‐digit number can be used to instantly access a given location setup. • An alphanumeric Name • TEL menu settings – This is a snapshot of all settings of the TEL menu. • AUDIO settings (optional) – This is a snapshot of all settings of the AUDIO menu. Note, the front panel Left/Right and Snd/Rcv mixer assignments on the Zephyr Xstream MX/MXP model’s front panel are included along with the settings in the AUDIO menu. Activating a Location Setup – Step by Step Note that this process is the same as accessing a Dial Setup, only a three‐digit number is used. 1. Press the <AUTO> key. 2. To choose a Location Setup you have two options: • Use the <5> and <6> to scroll through the list of 30 Location Setups. These are setups # 100 to 129 ‐ may need to press and hold <6> to scroll past the Dial Setups. • Use the Keypad to enter the three‐digit number for the desired setup. That setup will be displayed and will be highlighted. 3. Once the desired setup is highlighted, press the <AUTO> button to activate the configuration sequence. Once configuration is complete, the Xstream will show the following screen: Section 4: BASIC OPERATION & INSTALLATION 73 USER’S MANUAL ...
Page 86 4. Press <5> then <SEL> to reboot. The illustration below shows several sample Location Setups. Note that the notation “A” in the right column indicates that a particular setup includes AUDIO menu settings as well as TEL menu settings. Creating Location Setups – Set by Step 1. Configure the Zephyr Xstream with the ISDN line to be used at this location (see below). You must make certain that the menu selections in the TEL and AUDIO (optional) menus are set correctly: • All menu settings in the TEL menu. Ideally, you should be connected to the line for the target location, so you can verify the ISDN settings are correct. In that case, the line status should indicate Ready & Ready (with ETS 300 ISDN it is acceptable if Deactivated & Deactivated is displayed). If you are not connected to the line that's ok too, but you should be certain that the menu selections in the TEL menu are set correctly. NOTE: TEL menu settings are always included in a Location Setup. HOT TIP! You can activate an existing setup to recall these parameters, assuming you have an existing setup for the line in question. Do this first, and then make the necessary changes to the AUDIO menu settings.
Page 87 USER’S MANUAL 2. Press the <AUTO> key. Choose the Location Setup to be entered. Note: You can edit an existing setup by selecting it. To choose a Location Setup you have two options: • Press and hold the <6> to scroll to the list of 30 Location Setups (these are setups # 100 to 130). • Use the Keypad to enter the three‐digit number for the desired setup. That setup will be displayed and will be highlighted. 3. Once the desired setup is highlighted, press the <SEL> button to allow entry of (or editing of) the information for this setup. If an empty setup was selected, "Unused Entry" will be displayed (see below): 4. Press <SEL> to continue to the screen shown below: or press <5> and then <SEL> to escape without changing. 5. Enter a name (see the beginning of this chapter section for an example on how to select and enter information). 6. You can choose whether to Include Audio settings or not, by selecting this menu selection. The choices are Tel (no Audio menu information included with this Location setup) or Tel. & Audio. If Tel is selected the system will include the current settings in the TEL Menu with this setup. Each time this setup is activated, the system will reset the TEL Menu items to these setting before dialing the number(s) for this setup. If Tel. & Codec was selected the AUDIO settings will also be changed when this setup is activated. Section 4: BASIC OPERATION & INSTALLATION 75 ...
Page 88 6. Move the cursor to the word Save and then press <SEL> to complete the entry process. Editing Location Setups 1. If this setup includes audio information, configure the Zephyr Xstream’s AUDIO menu and front panel mixer assignments as it will be used when connected to this location. If the existing Audio information included in this setup is correct, simply activate the setup to recall those settings now (then drop the call (s) by pressing drop 4 times). If you are creating a Dial Setup that does not include audio settings, you can skip this step. 2. Configure the Zephyr Xstream with the information ISDN line to be used at this location (see below). You must make certain that the menu selections in the TEL and AUDIO (optional) menus are set correctly 3. Press the <AUTO> key. Choose the Location Setup to be edited. To choose a Location Setup you have two options: • Press and hold the <6> to scroll to the list of 30 Location Setups (these are setups # 100 to 130). • Use the Keypad to enter the three‐digit number for the desired setup. That setup will be displayed and will be highlighted. 4. Once the desired setup is highlighted, press the <SEL> button. The name of the setup will be displayed. Press <5> and then <SEL> to edit the setup. 5. Edit the name (see the beginning of this chapter section for an example on how to select and enter information). You can choose whether to Include Audio settings or not, by selecting this menu selection. The choices are Tel (no Audio menu information included with this Location setup) or Tel. & Audio. 6. Move the cursor to the word Save and then press <SEL> to complete the editing process. Section 4: BASIC OPERATION & INSTALLATION 76 USER’S MANUAL ...
Page 89: Codec Settings: Transmit, Receive, Bit Rate, & Sample Rate
Codec Settings: Transmit, Receive, Bit Rate, & Sample Rate When calling another ISDN codec (e.g. mode = Zephyr), these settings must be set correctly to achieve an end‐to‐end bi‐directional audio connection. Since this is just an introduction, we will just briefly tell you what each is. For more information, see Section 11 (detailed Menu Reference). Section 6 (Audio Coding Reference) covers the different transmission modes in detail, including the advantages and disadvantages of each. NOTE: When placing a call to a telephone (mode = Phone), or to a Zephyr Xport that is on a POTS line (mode = Xport) this settings can be ignored. Transmit This determines the format of the audio locally coded for transmission. In order for the far end to receive this audio, the far end codec must: 1) Support this transmission mode; and 2) Must have its "Receive" (or "decoder") mode set appropriately to receive the mode being sent. 3) The bit rate and sample rate settings must be matched between the two ends. COMPATIBILITY TIP! The AAC and AAC-LD standards are new, and are not supported by older codecs. When communicating with older equipment, you must make some other choice.
Page 90 If you choose not to use the "Auto" setting, you will need to know the Transmit (“Xmt” or “Encoder”) mode of the far end to determine the correct setting. HOT TIP! You can determine the correct setting for the Receive mode by first using the Auto mode. Once the Xstream gets a receive lock simply press the <CODEC> button and check what Receive mode was found. Then press <CODEC> again and change the Receive selection to this option.
Page 91 COMPATIBILITY TIP! The Zephyr Xstream specifies the bit rate per channel. Since certain modes (i.e. stereo or mono-128) require two channels to operate, the total (aggregate) bit rate would be double the number displayed in these cases. This is important as some other brands of codecs specify the total bit rate.
Page 92 IMPORTANT TIPS! 1) Zephyr Xstream only supports the 24kHz sample rate in MPEG Layer-2. Therefore, the Sample Rate menu does not have this option. The “L2 Half 64” Transmit and Receive modes are used to achieve 24kHz operation in MPEG Layer-2.
Page 93: Guided Tour Of The Menus
USER’S MANUAL GUIDED TOUR of the MENUS The Menu Tree This section gives an overview of the Xstream’s front panel menus suitable for the average operator. Section 11 is a detailed reference of the menus. We assume you have already read Section 4 (Basic Operation), which introduces the system and its basic operation. Note that most of the functions described in this section are also available in the HTTP configuration pages that you may access using your favorite web browser. Of course, you will have to give your Xstream a suitable IP address before you can access it in that manner. Where applicable, the web pages have been included in this section for your reference. Many settings are duplicated on the front panel and the web pages for maximum flexibility. First, let’s lay out for you the Xstream’s home page and the front panel menus we'll be discussing. The next few pages show each of the menus in detail. Note that we show the AUDIO menus twice, once for the Xstream, and again for the Xstream MX/MXP models. Web pages will also be shown to illustrate where you can find these settings in the web interface. Zephyr Xstream Home Page Section 5: GUIDED TOUR of the MENUS...
Page 94 MENUS - The CODEC, TEL & SYSTEM Menus CODEC CO DEC CODEC Compat. Zephyr CODEC S IP Port UDP Port HTTP Port TCP Port USER’S MANUAL (all versions) SYSTEM SYSTEM SYSTEM SYSTEM 5060 9150 8080 8800 SYSTEM SYSTEM Section 5: GUIDED TOUR of the MENUS SYSTEM SYSTEM SYSTEM...
Page 95 USER’S MANUAL MENUS - The DIAL & AUTO Menus (all versions) MANUAL DIALING ACTIVATING A DIAL SETUP DIAL Phon e # DIAL DIA L Phon e # DIAL DROP F our Times Section 5: GUIDED TOUR of the MENUS ACTIVATING A LO CATION SETUP AUTO 2 DIG ITS AUTO...
Page 96 USER’S MANUAL MENUS - The Audio Menu (non-mixer version) AUDIO Gain T rim Send Mix Discrete AUDIO AUDIO Section 5: GUIDED TOUR of the MENUS...
Page 97 USER’S MANUAL MENUS - The Audio Menu MX/MXP 1-4 Times 1-2 Times Assignm e nt Unlo ck e d Section 5: GUIDED TOUR of the MENUS (mixer versions) AUDIO AUDIO AUDIO AUDIO AUDIO...
Page 98: The Default Screen
MX/MXP models). To disable this screen, go to the 5th SYSTEM menu and set the Timeout option to "No". Audio Menu 5.3.1 The Zephyr Xstream's Audio Menu (non-mixer version) AUDIO - Screen 1 Pressing the AUDIO button once will show the first AUDIO menu, with the following selections. USER’S MANUAL Encoder &...
Page 99 (nominal). Lower operating levels can be accommodated using an external pad. Gain Trim This allows a fine adjustment of send level. This adjustment allows compatibility with a variety of equipment. This adjustment is in the digital domain and adjusts both the analog and AES/EBU send level. AUDIO - Screen 2 Pressing the AUDO Button a third time will show this screen with the following options. Input Source This allows selection of either the Analog, AES/EBU or Livewire input for your Send to Network Audio. AES Sample This option determines what source is used to clock the AES/EBU output. The Input selection clocks the AES/EBU output to the AES/EBU signal fed to the rear‐panel AES/EBU input and is the most commonly used option. The Sync selection clocks the AES/EBU output to the AES/EBU signal fed to the rear‐panel Sync‐in jack. See Section 11. (Detailed Menu Reference) Audio 5.3.2 Zephyr Xstream's MX & MXP’s Menu Don't forget that you can create Location Setups to easily recall the menu selections of the AUDIO menus. Even if the ISDN line is the same, these Location Setups can be used to program these parameters (i.e. for different shows or uses in a dedicated situation. In the cases where the ISND settings have not changed, just choose CANCEL when requested to reboot. Section 5: GUIDED TOUR of the MENUS...
Page 100 AUDIO - Screen 1 (Status) The first AUDIO menu has the individual levels for each of the Main Mixer inputs. These levels are post fader and therefore can be used to view relative levels of each source in the final mix. AUDIO - Screens 2 - 5 The second, third, fourth and fifth AUDIO menus allow adjustments of parameters for Main Mixer inputs 1‐4, respectively. These screens are identical and have the following information: Ch “x” Gain This menu selection allows you to select from four different gain settings (2 mic. and 2 line) for Main Mixer input x. The options for this setting are: MIC +44 : ‐55 dBu nom. MIC +30: ‐40 dBu nom. Line Con: ‐10 dBu nom. Line Pro: +4 dBu nom. The nominal values given above assume the fader is mid‐position and the AGC/Limiter (processor) is off Ch “x” Pan This menu selection is a "panorama" adjustment that allows you to vary the amount of audio sent to the Left and Right codec inputs. Note that this adjustment has no effect unless the input x has been assigned to both A and B using the front panel control. Ch “x” Proc This menu selection allows you to enable the built in AGC/Limiter Main Mixer input x. You will be able to see the level for input "x" as well as the AGC and Limiter working by looking at the small bar‐graphs and gate indicator to the right side of this screen. The built in Omnia processor is sophisticated and offers the following options: None: ...
Page 101 This disables the input processor for channel "x". Voice L: (Voice Low) This is the least aggressive setting, intended to protect against minor level problems when running a live show. It provides AGC with no limiting. Ideally where you want the most natural sound in live situations where mixing could be difficult. Voice M: (Voice Medium) This is a moderate setting. It combines AGC with a small amount of limiting. This provides good dynamics control, and also serves to "beef up" the voice. This setting is the most frequently used general purpose option. Do not use this setting for music. Voice H: (Voice High) This is an aggressive setting employing both AGC and limiting. This setting is perfect for easily excited talent such as sportscasters. Do not use this setting for music. Music: This setting offers a slow AGC only. Low Cut This is a low cut filter designed to reduce wind noise, pops and breath noise. This can be enabled independently for each input. AUDIO - Screens 6 & 7 These screens are used to set up Local Mix 1 (screen 6) and Local Mix 2 (screen 7). The following options allow the operator to customize the Local Mixes as required for different applications are as needed to accommodate talents' needs. The following selections are displayed are offered. Mon “x” Balance This adjustment allows you to vary the ratio of Send to Receive audio in Local Mix x. This control has no effect if both SND and RCV have not been selected on the front panel. USER’S MANUAL & Section 5: GUIDED TOUR of the MENUS...
Page 102 Mon “x” Rcv Mix This parameter, in combination with Mon x Rcv Pan selection (see below) is useful for when only a single return channel is being sent from the far end. The default is stereo. The options are: Stereo: This choice routes the incoming (far end) audio to the headphones as received. The A channel audio appears on the Left side of the Local Mix and the B channel audio appears in the Right side of the Local Mix. Left: This option routes the incoming channel A (Left) audio to both sides of the Local Mix. Right: This option routes the incoming channel B (Right) audio to both sides of the Local Mix. Left +Right: This option sums (i.e. mixes) the incoming A & B audio channels and routes this to both sides of the Local Mix. Particularly useful when the both a mix minus and a producer IFB (interruptible foldback) are being recieved as it allows both feeds to be heard in both ears. Mon “x” Rcv Pan Whenever Rcv is selected using the front panel button, this control determines the ratio of the amount of the audio fed to the A (Left) and B (Right) channels of the Local Mix. Mon “x” Xmt Mix This option operates in the same way as Mon "x" Rcv, only the send audio is used. This is particularly useful when creating two separate mono mixes to be sent to two far end locations as it can allow the talent to hear the correct return mix in both ears. The options are: Stereo: This choice routes the outgoing (near end) audio to the headphones just as transmitted by the codec section. The A channel audio appears on the Left side of the Local Mix and the B channel audio appears in the Right side of the Local Mix. Left: This option routes the outgoing channel A (Left) audio to both sides of the Local ...
Page 103 Right: This option routes the outgoing channel B (Right) audio to both sides of the Local Mix. Left +Right: This option sums (i.e. mixes) the incoming A & B audio channels and routes this to both sides of the Local Mix. Mon “x” Xmt Pan Whenever Xmt is selected using the front panel button, this control determines the ratio of the amount of the audio fed to the A (Left) and B (Right) channels of the Local Mix. AUDIO - Menu 8 This menu has three options: Phantom This option allows you to disable (Off) or enable (On) phantom power for inputs 1 & 2. Phantom power is the full 48 volts and is fed to both inputs 1 & 2. Level Out This option sets the nominal output level for the line outputs for Monitor Mix 2. The options are "Professional" (+4 dBu nominal) or "Consumer" (‐10 dBu nominal). Assignment This setting this to Locked locks the input assignments (A/B select) and monitor assignments (Send/Receive select) to their current settings. USER’S MANUAL Section 5: GUIDED TOUR of the MENUS...
Page 104 Livewire Since Livewire is one way of getting audio to and from your Xstream, we will show those menus here as well. Please refer to Section 8 for details on configuration of Livewire. These menus enable you to specify the Livewire streams being created by the codec (“sources” delivered to the Livewire network) and the audio streams being delivered by the codec (“destinations” from the Livewire network). Expert mode enables you to specify custom backfeeds if required. Please see Section 8 for details on Livewire configuration. Livewire – Basic Setup Livewire – Expert Setup USER’S MANUAL Section 5: GUIDED TOUR of the MENUS ...
Page 105: The Codec Menu
USER’S MANUAL Codec Menu The settings of the codec menu will determine what codecs with which you can communicate, as well as the quality and delay of the coded audio stream. For more on these selections see Section 6 (Audio Coding Reference) and Section 10 (Applications Guide). Note that the menus shown here are applicable when the Interface setting in the TEL Menu is set to ISDN or V.35 split. If V.35 single or Ethernet is selected for Interface, the CODEC menus will be slightly different. See Appendix 6 (Special Operational Modes) for more details. CODEC - Screen 1 (Status) The first press of the <CODEC> button shows a screen with a summary of the codec settings (see below) Section 5: GUIDED TOUR of the MENUS...
Page 106 CODEC - Screen 2 A second press of the <CODEC> button shows the screen above: On this screen are the settings that affect compatibility with far‐end codecs. These parameters also affect the quality (and delay) of the audio transmitted. Transmit This determines the format of the audio locally coded for transmission. In order for the far end to receive this audio, the far end codec must: 1) Support this mode; and 2) Must have its Receive (or decoder) mode set appropriately to receive the mode being sent. 3) The bit rate and sample rate settings must be matched between the two ends. USER’S MANUAL Section 5: GUIDED TOUR of the MENUS...
Page 107 IMPORTANT TIP! When referring to Transmit modes we use the Term "Mono 64" to refer to a mono signal using a single B channel. Therefore, the audio stream may be at 56 or 64kbps since the call could be placed at either rate. Similarly, we use the term "Mono 128"...
Page 108 This parameter must be matched between the Zephyr Xstream and the far end codec. COMPATIBILITY TIP! The Zephyr Xstream specifies the bit rate per ISDN or V.35/X.21 channel. Since certain modes (i.e. stereo or mono-128) require two channels to operate, the total (aggregate) bit rate would be double the number displayed in these cases.
Page 109 IMPORTANT TIP! 1) Zephyr Xstream only supports the 24kHz sample rate in MPEG Layer-2 (or when used in “IP streaming mode” -See Appendix 6 - Special Operational Modes for more on that mode of operation). Therefore, the Sample Rate menu does not have this option. The “L2 Half 64” Transmit and Receive modes are used to achieve 24kHz operation in MPEG Layer-2.
Page 110 COMPATIBILITY TIP! Not all codecs support all sample rates. 32kHz is most commonly supported in MPEG Layer- 3, whereas 48kHz is more commonly used in Layer-2. 24kHz is sometimes seen in Layer-2. In this case, set your Transmit or Receive to “L2 Half/24”...
Page 111: The Tel Menu
Menu TEL - Screen 1 (Status) The first press of the TEL button will show the Tel Status screen. This gives the current status of each of your ISDN B channels ("lines" 1 & 2). The most commonly seen status words are the following (See Section 10 for additional information): Ready – This indicates that this B channel has correctly initialized with the Telco and is able to make and receive calls. The line is "idle". ISDN TIP! A false ready may be displayed if your Telco setting is incorrect (see below), or if SPIDs are not entered when required. If your ISDN circuit uses the National ISDN or DMS-Custom ISDN protocol, you must enter Directory Numbers.
Page 112 SPID Pending – This may be displayed if an external NT1 has not completed initializing to the line. Or, it may indicate that the internal NT1 has recognized the line and is initializing to the rest of the Xstream (i.e. slowly blinking NT1 LED). Finally, we have seen this when the ISDN circuit is grossly misconfigured. SPID Error – This indicates that the SPID for this number (located in the 2nd TEL menu) is incorrect. SPIDs are only needed with certain ISDN protocols used in the USA & Canada. SPIDs are not needed for ETS‐300 (Euro ISDN), INS 64 (Japanese ISDN) or AT&T Custom Point to Point. Inactive – This indicates that internal NT1 is not active nor is there an active S interface. This may mean that the ISDN line is not connected or is faulty. NOTE that Euro ISDN (ETS 300) lines are permitted to go into this state, and will act normally when the time comes to dial. TEL - Screen 2 Pressing the TEL button a second time, shows this screen. These important parameters must be entered about your ISDN circuit in order for it to properly function. If your ISDN BRI circuit protocol is ETS 300 (Euro ISDN), INS 64 (Japanese ISDN), or AT&T Cust PTP you can leave all of these blank. SPID 1 & SPID 2 Users in the USA & Canada should enter their SPIDs here. SPID mistakes (or confusion) are among the most common problems users in these countries have. Here are some tips to assist you. ISDN TIP! 1. Incorrectly entered SPIDs are the most common problem users in the USA & Canada have placing ISDN calls.
Page 113 We've included numerous tips on SPIDs and the TELCO setting in Section 11.5 (Detailed Menu Reference). MSN/DN 1 & MSN/DN 2 DNs (Directory Numbers) are needed in the USA & Canada, when the ISDN protocol is National ISDN or DMS‐Custom. These are simply your 7‐digit telephone numbers, as would be found in the telephone directory. MSNs (Multiple Subscriber Numbers) are used with ETS 300 (Euro ISDN) and are optional. They can be used if you are sharing your BRI circuit with other equipment. For more information, see Section 11 (Detailed Menu Reference). HOT TIP! The Directory Numbers or DNs (sometimes called Listed Directory Numbers or LDNs) are your 7-digit phone number as would be found in the Telephone Directory. They normally do not include the area code.
Page 114 TEL - Screen 3 Pressing the TEL a third time will show this menu. The options for this menu are as follows: Telco This is the protocol for your ISDN circuit. Outside of the USA & Canada, it is usually ETS 300 (Euro ISDN). In the USA & Canada, the most common protocol is National ISDN 1 (use the Natl.I‐ 1 setting). In Japan it will be INS 65 (Japanese ISDN). IMPORTANT! Each ISDN circuit has a certain ISDN protocol (or “switch type”) that the phone company should have given you. By far the most common protocols in the USA & Canada are National ISDN-1 &...
Page 115 NOTE: Call from a Zephyr Xport that is calling from a POTS (analog) line is considered a Phone mode call. Interface This allows you to choose between the ISDN, Ethernet and, if you have the optional V.35/X.21 interface, V.35. To use the V.35/X.21 interface choose V.35 Split or V.35 Single. To use the built‐in ISDN interface set this to ISDN. To stream IP over the Ethernet choose Ethernet. HOT TIP! Choosing Ethernet or V.35 Single will change the available options for the CODEC and DIAL menus. See Appendix 7 (Special Operational Modes) for additional information. Prefix This will generally be left blank. Any number(s) entered in this field will be appended to the beginning of any number entered when dialing (either manually or by using Dial Setups). The most common use of this is when your ISDN circuit requires that a "9", "8", or other digit be used before dialing a number. Since your Xstream may be moved from line to line, and not all lines may need this prefix, we allow you to enter this prefix here, so you can omit it from Dial Setups. In this way the same Dial Setups can be used on different lines. ...
Page 116 UDP Port After the a SIP session is initiated, or when using RTP streaming mode, the actual audio data is carried by UDP packets sent to and received on this port. HTTP Port The Xstream is able to use HTTP protocol to send an audio stream to a PC‐based player for local debugging. At this time, these software players only support Layer 3 streaming. For stability, this should only be used for debugging, not on‐air delivery or monitoring. TCP Port This setting is currently unused. TEL - Screen 5 The fifth TEL screen has various settings for fine‐tuning ISDN and v.35 performance. DTMF Enables or disables sending DTMF tones over voice calls using the front panel keypad. We offer the option of turning this off in the case that you may be entering information with the keyboard while dialed up on a voice call Redial Sets the delay between automatic ISDN redials, or disables the feature. If an outgoing call is active when the Xstream loses ISDN connectivity or power, it will attempt to re‐ establish the call as soon as the error state is cleared (that is, when the ISDN comes back to life, or power is restored to the Xstream). You can adjust the time between these attempts, or disable them entirely. The redialing feature is also used when attempting to dial via contact closures and v.35 roll‐over (see next item). ISDN Backup On an Xstream with both a v.35/x.21 card and an ISDN card, you can enable a rollover in case of v.35/x.21 failure. If you choose to enable this feature, this menu item will allow you to select an dial setup (any of your first ten) to be used. Note that by enabling this feature, you also allow ISDN calls to switch the Xstream from v.35/x.21 mode to ISDN mode (This is necessary because while one side of the high speed serial connection may see a failure, the other side could be operating as usual. In this case, when the side seeing the failure condition switches to ISDN mode and redials, the other side must respond to the ISDN request). USER’S MANUAL . ...
Page 117: The System Menu
USER’S MANUAL System Menu The system menu has many selections, a number of which the average user need not consider. We'll cover those of interest to the average user in this section. See Section 11 (Detailed Menu Reference) for more information on the rest. SYSTEM - Screen 1 (Status) This menu gives summary information about the system. It gives a quick summary of the information customer support will need to help you. Should you need to call us, have this information ready. Uptime – This is the length of time since the last reboot (power cycle or warm boot). This can be helpful when troubleshooting intermittent problems. The format is D‐HH:MM:SS. D = Days; H = hours; M = Minutes; S = Seconds. Loopmode – This is the state of the Loopmode setting in the 5th SYSTEM menu. This should normally be set to none. Version – This is the software version for your Zephyr Xstream. When you call Telos Technical Support they may request this information. If version number ends in 'b' or 'r' and the unit has a 100Base‐T Ethernet jack on the back, it is capable of Livewire streaming. Section 5: GUIDED TOUR of the MENUS...
Page 118 USER’S MANUAL Interface – This is the current setting of the Interface option located on the third TEL menu. SYSTEM - Screen 2 This menu has the most frequently used System menu items. These are: Contrast This allows you to change the contrast of Xstream's LCD display to accommodate differing room lighting or viewing angle. Volume No, this doesn't belong in the AUDIO menu... this volume selection is to adjust the volume of the different sounds the Xstream makes when keys are pressed, calls connect, or other things happen. rs232 This allows you to easily set the bit rate of the Zephyr Xstream's RS‐232 port on the rear panel. About This gives you information about your Xstream. SYSTEM - Screen 3 Pressing the SYSTEM button a third time shows this screen with the selections needed to configure your IP (Internet Protocol) settings. You won't need to enter this to operate the unit from the front panel. You will need to enter this information if you will be connecting to the Xstream's Ethernet port (i.e. if you are using Livewire, will be using the Ethernet streaming mode, or will be controlling the unit remotely using a web browser). You will also need to enter these items when the time comes to update the Xstream's operating software. For more information, see Section 11 (Detailed Menu Reference) and Section 13 (Deep Tech & Troubleshooting Info.) Section 5: GUIDED TOUR of the MENUS...
Page 119 USER’S MANUAL SYSTEM - Screen 4 Current Version Legacy Versions The fourth press of the SYSTEM button brings you to the reboot or FTP screen used to update the Zephyr Xstream's software. Older versions of the software use an FTP site for updates. Current versions are updated via their web page interface. For more information, see Sections 11 (Detailed Menu Reference) and 13 (Deep Tech & Troubleshooting Info.) Reboot This option is used to restart the Xstream without cycling power. It can be handy. The Xstream is designed for long term operation without user intervention. If you frequently experience problems that seem to be solved by rebooting, be sure to contact Telos with all relevant details, so we can assist you in determining the cause. SYSTEM - Screen 5 The fifth SYSTEM menu screen has several options, as follows: Loopmode This allows you to activate various loop modes used to troubleshoot the unit and its environment. This should normally be set to None, and will revert to that value on startup. See Section 13 (Deep Tech & Troubleshooting) for troubleshooting information. Timeout This determines whether the system will return to the default screen after a period of inactivity. The default setting is "Yes" meaning that the system will return to the default menu after 120 seconds of inactivity. Ancillary See Section 11 (Detailed Menu Reference) for details on this option. The usual setting is "On" which is the default. Section 5: GUIDED TOUR of the MENUS...
Page 120: Auto, Dial & Drop
USER’S MANUAL Restore Factory Defaults... This selection allows you to clear all settings of your Xstream. All menu options will be returned to the factory default settings. Don't use this option unless you are sure you wish to erase these entries. However, Dial Setups and Location Setups will not be erased. You will be prompted with a WARNING screen inquiring if you really wish to erase all your settings. Press the <SEL> button (or any of the MENU buttons) if you wish to abort. If you choose to proceed with erasing your settings, select OK by pressing <6> or <5>to select OK. Then press <SEL> to proceed. In rare cases where the Zephyr Xstream's internal memory has become corrupted, the Restore factory defaults selection may help clear up the misbehavior of the unit. Auto, Dial & Drop These functions are covered in detail in Section 4.2 – Intro to the Xstream’s Menu System and in Section 11.7 – The Works, Detailed Menu Reference. Please refer to those sections for more information. Section 5: GUIDED TOUR of the MENUS...
Page 121: Audio Coding Reference
AUDIO CODING REFERENCE Introduction to Audio Coding Technology Introduction Audio takes up a lot of data. Just a regular phone call uses 64,000 bits per second. Without data reduction, CD‐quality quality audio — 16 bits at 44.1kHz sample rate — requires a transmission capacity of about 706 thousand bits per second (kbps) for each audio channel. But, the wires we use for remote broadcasting are on a telephone system designed for voice‐grade communications: 8 bits at 8kHz sample rate, or 64 thousand bits per second (kbps) per channel. That’s 11% of what we need. CURIOSITY NOTE! You can arrive at these same numbers with nothing more complicated than grade-school math. Just multiply the sample rate by the sample depth: 44,100 samples per second * 16 bits per sample = 705,600 bits per second for CD-quality mono audio.
Page 122 USER’S MANUAL delta, between successive audio samples compared to using the individual values. Further efficiency is had by adaptively varying the difference comparator according to the nature of the program material. G.722 and APT‐X are examples of ADPCM schemes. They achieve around a factor of 4:1 reduction in bitrate. G.722 achieves additional efficiency by allocating its bits to match the patterns in the human voice, and it’s considered adequate for news and talk programming over ISDN. But, for high‐ fidelity transmission, algorithms with more power are required. These are based on psychoacoustics, where the coding process is adapted to the way we hear sounds. There are several algorithms available, with varying complexity and performance levels. Some years ago, the international standards group ISO/IEC established the ISO/MPEG (Moving Pictures Expert Group), to develop a universal standard for encoding moving pictures and sound for digital storage and transmission media. The standard was finalized in November 1992 with three related algorithms, called Layers, defined to take advantage of psychoacoustic effects when coding audio. Layer 1 and 2 are intended for compression factors of about 4:1 and 6 or 8:1 respectively, and these algorithms have become popular in satellite and hard‐disk systems. Layer‐3 achieves compression up to 12.5:1 — 8% of the original size — making it ideal for ISDN. Basic Principles of Perceptual Coding With perceptual coding, only information that can be perceived by the human auditory system is retained. Lossless – which, for audio, translates to noiseless – coding with perfect reconstruction would be an optimum system, since no information would be lost or altered. It might seem that lossless, redundancy‐reducing methods (such as PKZIP, Stuffit, Stacker, and others used for computer hard‐disk compression) would be applicable to audio. Unfortunately, no constant compression rate is possible due to signal‐dependent variations in redundancy. There are highly redundant signals like constant sine tones (where the only information necessary is the frequency, phase, amplitude, and duration of the tone), while other signals, such as those which approach broadband noise, may be completely unpredictable and contain no redundancy at all. Furthermore, looking for redundancy can take time. While a popular song might have three choruses with identical audio data that would need to be coded only once, you’d have to store and analyze the entire song in order to find them. Any system intended for a real‐time use over telephone channels must have a consistent output rate and be able to accommodate the worst case, so effective audio compression is impossible with redundancy reduction alone. Fortunately, psychoacoustics permits a clever solution! Effects called “masking” have been ...
Page 123 USER’S MANUAL ear’s sensitivity to signals near the constant tone is greatly reduced. Tones that were previously audible become “masked” in the presence of “masking tones,” in this case, the one at 300 Hz. All signals below the upper “threshold of audibility” curve, or Masking Threshold are not audible, so we can drop them out or quantize them crudely with the least number of bits. Any noise which results from crude quantization will not be audible if it occurs below the threshold of masking. The masking depends upon the frequency, the level, and the spectral distribution of both the masker and the masked sounds. Masking effects in the frequency domain. A masking signal inhibits audibility of signals adjacent in frequency and below the threshold. To benefit from the masking effects, perceptual coders use a filter bank to divide the input audio into multiple bands for analysis and processing. The maximum masked noise level is calculated depending upon the spectral content, and the available bits are allocated so as to keep the quantization noise below the masking threshold at every point in the spectrum. While coding efficiency increases with more bands and better frequency resolution, the time domain resolution decreases simultaneously owing to an inevitable side‐effect of the band filtering process. Higher frequency resolution requires a longer time window – which limits the time resolution. Happily, masking works also in the time domain. A short time before and a longer time after a tone is switched on and off, other signals below a threshold amplitude level are not noticeable. Filter banks with higher frequency resolution naturally exploit the ear’s time‐masking properties. Section 6: AUDIO CODING REFERENCE 111...
Page 124 USER’S MANUAL Masking effects in the time domain. Masking occurs both before and after the masking signal. The combined results of time and frequency vs. masking. Signals under the curve are inaudible. IMPORTANT! Due to a perceptual coder’s reliance on precisely modeling principles of human perception, audio to be coded should not be processed with any non-linear dynamics-processing such as clipping, multi-band compression, or hard limiting.
Page 125 The steps involved in the perceptual coding process are shown below: The components work as follows: • The analysis filter bank divides the audio into spectral components. Sufficient frequency resolution must be used in order to exceed the width of the ear's critical bands, which is 100 Hz below 500 Hz and 20% of the center frequency at higher frequencies. • The estimation of masked threshold section is where the human ear/brain system is modeled. This determines the masking curve, under which noise must fall. • The audio is reduced to a lower bit rate in the quantization and coding section. One the one hand, the quantization must be sufficiently coarse in order not to exceed the target bit rate. On the other hand, the error must be shaped to be under the limits set by the masking curve. • The quantized values are joined in the bit stream multiplex, along with any side information. Doing audio coding effectively means managing several tradeoffs. Most important is the number of samples coded together in one frame. Long frames have high delay, but are more efficient because the header and side information is transmitted less frequently. Longer frames offer the possibility to use filter banks with better frequency resolution. A fundamental principle in signal processing is that spectral splitting filters may have either good time resolution, or good frequency resolution, but not both. This makes sense when you consider that a longer time window means that the analyzer has more complete information, more full audio cycles, to work with . In the case of rapidly changing input signals (transients) long frames are poorer than short ones because the time spread will lead to so called pre‐echoes. For such signals, the size of the frame should correspond to the temporal resolution of the human ear. This can be achieved by using short frames or by changing the frame length according to the immediate characteristics of the signal. Perhaps this is the DSP designer’s equivalent to the economist’s TANSTAAFL: There ain’t no such thing as a free lunch.
Page 126 The MPEG process is open and competitive. A committee of industry representatives and researchers meet to determine goals for target bit rate, quality levels, application areas, testing procedures, etc. Interested organizations that have something to contribute are invited to submit their best work. A careful double blind listening test series is then conducted to determine which of the entrant's technologies delivers the highest performance. The subjective listening evaluations are done at various volunteer organizations around the world that have access to both experienced and inexperienced test subjects. Broadcasters are the most common participants with many of the important test series conducted at the BBC in England, the CBC in Canada, and NHK in Japan. Finally, results are tabulated, a report is drafted and ultimately a standard is issued. In 1992, under MPEG1 (the first of the MPEG standards), this process resulted in the selection of three related audio coding methods, each targeted to different bit rates and applications. These are the famous layers: 1, 2 and 3. As the layer number goes up, so does performance and implementation complexity. Layer 1 is not much used. Layer‐2 is widely used for DAB in Europe, audio for video, and broadcast playout systems. Layer‐3 – which Telos was the first to use in the Zephyr – is widely used in broadcast codecs and has gone on to significant Internet and consumer electronics fame under the moniker derived from the file extension: MP3. MPEG2 opened the door for new work, and some minor improvements were added to both Layers 2 and 3. In 1997, the first in the AAC family was added to the MPEG2 standard. MPEG4 is ongoing now, but it has already been decided that AAC will be the “general audio” coder under this umbrella. (MPEG3 was skipped for reasons unknown.) MPEG2 AAC (Advanced Audio Coding) The MPEG2 AAC system is the newest audio coding method selected by MPEG and became an International standard in April 1997. It is a fully state‐of‐the‐art audio compression tool that provides performance superior to any known approach at bit rates greater than 64 kbps and excellent performance relative to the alternatives at bit rates reaching as low as 16 kbps. The idea that led to AAC was not only to start fresh, but also to combine the best work from the world’s leading audio coding laboratories. Fraunhofer, Dolby, Sony, and AT&T were the primary collaborators that offered components for AAC. The hoped for result was ITU (International Tele‐communications Union) “indistinguishable quality “at 64 kbps per mono channel. That is, Section 6: AUDIO CODING REFERENCE 114...
Page 127 quality indistinguishable from the original, with no audio test item falling below the “perceptible, but not annoying” threshold in controlled listening tests. The MPEG test items include the most difficult audio known to codec researchers, so this was daunting challenge. The thinking was that if a codec could pass this test, it would surely be transparent for normal program material like voice and pop music, which are much easier to encode. AAC designers chose to use a new modular approach for the project, with components being plugged‐in to a general framework in order to match specific application requirements and the always‐present performance/complexity/delay tradeoffs. Compared to the previous layers, AAC takes advantage of such new tools as temporal noise shaping, backward adaptive linear prediction and enhanced joint stereo coding techniques. AAC supports a wide range of sampling rates (8–96 kHz), bit rates (16–576 kbps) and from one to 48 audio channels. AAC is a lot more sophisticated than the previous MPEG layers 2 &3, providing significantly more coding power. Because it uses a modular approach, new ideas can be developed and plugged in to the basic structure. This had the additional advantage that it was possible to combine the various components from different developers, taking the best pieces from each. AAC was built on a similar structure to Layer‐3, and thus retains most of its features. Nevertheless, compared to the previous MPEG layers, AAC benefits from some important new additions to the coding toolkit: • An improved filter bank with a frequency resolution of 2048 spectral components, nearly four times the number used by Layer‐3. • Temporal Noise Shaping, a new and powerful element that minimizes the effect of temporal spread. This benefits voice signals, in particular. • A Prediction Module guides the quantizer to very effective coding when there is a noticeable signal pattern, like high tonality. • Perceptual Noise Shaping allows a finer control of quantization resolution, so bits can be used more efficiently. Because it uses this modular approach, an implementer may pick and choose among the component tools to make a product with appropriate performance/complexity ratios. Or, new modules can be developed later and "plugged in" to its basic structure. Three default profiles have been defined, using different combinations of the available tools: • Main Profile. Uses all tools except the gain control module. Provides the highest quality for applications where the amount of random accessory memory (RAM) needed is not constrained. •...
Page 128 The block diagram of the AAC encoder is shown below. It is considerably more sophisticated than the previous MPEG Layer‐2 and Layer‐3 systems, and therefore offers more coding power. Because AAC was built on a similar structure to Layer‐3, it therefore retains some of its powerful features: • Redundancy Reduction. A Huffman encoding process causes values that appear more frequently to be coded with shorter words, while values that appear only rarely are coded with longer words. This results in an overall increase in coding efficiency – with no degradation, since it is a completely lossless process. • Bit Reservoir buffering. Often, there are some critical parts in a piece of music that cannot be encoded at a given data rate without audible noise. These sequences require a higher data rate to avoid artifacts. On the other hand, some signals are easy to code. If a frame is easy, then the unused bits are put into a reservoir buffer. When a frame comes along that needs more than the average amount of bits, the reservoir is tapped for extra capacity. • Ancillary Data. The bit reservoir buffer offers an effective solution for the inclusion of such ancillary data as text or control signaling. The data is held in a separate buffer and gated onto the output bit stream using some of the bits allocated for the reservoir buffer when they are not required for audio. • The Joint Stereo mode takes advantage of the redundancy in stereo program material. The encoder switches from discrete L/ R to a matrixed L+R/ L‐R mode dynamically, depending upon the program material. USER’S MANUAL Section 6: AUDIO CODING REFERENCE 116...
Page 129 Affordable, transparent, audio transmission for AM/FM radio or television audio. COMPTIBILITY TIP! The Zephyr Xstream cannot receive two J-stereo 64 feeds from the far end (after all, where would the audio be output?). When connected to two different sites simultaneously you can receive an audio feed back from each using G.722 or L3 Dual/Mono.
Page 130 USER’S MANUAL AAC-LD (AAC Low Delay) When announcers use codecs for broadcast remote applications, they often need to have natural two‐way interaction with other program participants located back at the studio, or callers. Because it is a hot topic for engineers working in the field of Internet telephony, a number of studies have been conducted to determine user ’s reactions to delays in telephone conversations. The data apply directly to the application of professional codecs to remotes, so it is interesting to take a peak over the shoulder of the telecom guys to see what they have learned. For broadcast remotes, we try to arrange our system so that there is no path for the field announcer’s voice to return to his/her headphones. Nevertheless, sometimes echo is unavoidable. For example, this can happen when a telephone hybrid has leakage or when a studio announcer has open‐air headphones turned‐up loud and the audio makes its way into the studio microphone. When there is no echo, it has been discovered that anything less than 100 ms one‐way delay permits normal interaction between participants. Between 100 and 250 ms is considered “acceptable.” ITU‐T standard G.114 recommends 150 ms as the maximum for “good” interactivity. Echo introduces a different case. As you might expect, echo is more or less annoying depending upon both the length of time it is delayed and its level. Telephone researchers have measured and quantified reactions, and ITU‐T G.131 reports the findings and makes recommendations. Summary of ITU-T G.13, with recommendations for designers of telephone systems that must cope with echo. This shows Talker Echo Loudness Rating vs delay. ...
Page 131 codec? Until recently, the answer was no, but new developments in codecs have changed the picture. One of the main objectives in audio coding is to provide the best tradeoff between quality and bit rate. In general, this goal can only be achieved at the cost of a certain coding delay. Codecs for voice telephone applications have use ADPCM and CELP because they have much lower delay than perceptual codecs. These are optimized for voice and can have reasonably good performance. Zephyr users have known for years that Layer‐3 offers all the fidelity needed for most broadcast situations. However, they also know that the delay of Layer‐3 can be frustrating, particularly if high fidelity is needed in both directions and parties at the two ends must carry on a conversation. The folks at Fraunhofer were aware of these factors, and have developed an extension to AAC called "AAC Low Delay," or "AAC‐LD" for short. ACC‐LD offers quality equivalent to Layer‐3 with less than 25% of the delay! AAC‐LD combines the advantages of perceptual coders (such as Layer‐3) with certain principles of speech coders. Compared to speech coders, AAC‐LD handles both speech and music with good quality. Unlike speech coders, however, audio quality scales up with bit rate, and transparent quality can be achieved. AAC‐LD’s coding power is roughly the same as Layer 3, meaning that mono high fidelity 15 kHz audio may be sent via one ISDN channel. With ISDN’s two channels, you achieve near CD quality stereo. Delay in perceptual codecs is dependent on several parameters: • Frame length. Time is required to collect all the samples for a frame. The longer the frame, the more the delay. • Filter bank delay. This causes an additional delay equivalent in time to the frame delay. • Look‐ahead delay for block switching. Layer‐3 and AAC use filter banks with high frequency resolution. For signals with high tonality, efficiency is high. But when there are transients, a dynamic switching process changes to a filter bank with lower frequency resolution and better time resolution. In order to correctly decide when to make this change, a look ahead process is required, adding delay. • Bit reservoir. The length of this buffer determines how much delay this process contributes. The overall delay is a combination of all of these components, divided by the sampling rate. The delay scales linearly and inversely with the sampling frequency. How AAC‐LD Gets its Low Delay AAC‐LD is based on the core AAC work, so much is similar, but each of the contributors to the ...
Page 132 • No dynamic block switching is used because the required look‐ahead delay is too big. The temporal problem that causes pre‐echoes is handled by the Temporal Noise Shaping module. • The “window shape” of the spectral filter is enhanced to be adaptive. Normally, the shape is a simple sine curve, but AAC‐LD can use a shape that has a lower overlap between the bands. This significantly improves performance with transients, without adding any delay. Amazing Performance Low delay would not be useful if the quality was not acceptable, of course. So how does AAC‐LD stack‐up? The results are shown below: AAC‐LD is clearly better than Layer‐3 for half of the test items, and as good for the remaining half. Not bad considering that AAC‐LD has a fraction of the delay. Performance comparison of Layer 3 and AAC-LD. A = AAC-LD, B = Layer 3. Results here are very good for AAC‐LD. Test items es01‐03 are speech. Since most mono remote broadcasts are speech, you can expect audio quality even better than with the familiar – and already perfectly useful – Layer‐3. Pop music is the test item sc03 and performance is good here as well. The one slight trouble spot is the glockenspiel at sm02… We expect you will find this mode very useful. ISO/MPEG LAYER-3 After extensive testing by broadcasting organizations around the world under the direction of the CCIR, it had been designated as the most powerful of the three audio coding systems USER’S MANUAL Section 6: AUDIO CODING REFERENCE 120...
Page 133 USER’S MANUAL standardized in ISO/MPEG IS‐11172. Prior to the availability of AAC, Layer‐3 was specifically recommended for 56 and 64 kbps channel. Layer-3 Features Psychoacoustic Masking The audio in Layer‐3 is divided into 576 frequency bands. First, a polyphase filter bank performs a division into the 32 “main” bands which correspond in frequency to those used by the less complex Layer‐2. Filters are then used to further subdivide each of the main bands into 18 more. At a 32 kHz sampling rate, the resulting bandwidth is 27.78 Hz – allowing very accurate calculation of the masking threshold values. Sufficient frequency resolution is available to exceed the width of the ear’s critical bands (100 Hz below 500 Hz; 20% of the center frequency at higher frequencies) across the audible spectrum, resulting in better hiding of noise than would otherwise be possible. Redundancy Reduction Redundancy reduction is accomplished by a Huffman coding process to take advantage of the statistical properties of the simplified signal output from the psychoacoustic stage. Values that appear more frequently are coded with shorter words, whereas values that appear only rarely are coded with longer words. This results in an overall decrease in the data rate, with no degradation, since it is a lossless reduction scheme. Notice that this redundancy reduction process is the ideal supplement to psychoacoustic masking. In general, maskers with high tonality have more redundancy but allow less masking, while noise‐like signals have low redundancy and high masking effect. Bit Reservoir Buffering Often, there are some critical parts in a piece of music that cannot be encoded at a given data rate without audible noise. These sequences require a higher data rate to avoid artifacts. Layer‐ 3 uses a short time “bit reservoir” buffer to address that need. Similar to a savings account, this buffer is filled in “easy times” with data bits that are not required for the actual frame. If a critical part occurs, the encoder can use the saved bits to code this part with a higher data rate. Ancillary Data The bit reservoir buffer offers an interesting capability: an effective solution for the inclusion of such ancillary data as text or control signaling. The data is held in a separate buffer and gated onto the output bitstream using the bits allocated for the reservoir buffer when they are not required for audio. Layer‐3 J‐Stereo A joint stereo mode permits advantage to be taken from the redundancy in stereo program material. The encoder switches from discrete L/R to a matrixed L+R/L‐R mode dynamically, depending upon the program content. The matrixed mode of operation takes of advantage of the usual redundancy of the “center” channel information and therefore significantly improves overall fidelity. ...
Page 134 HOT TIP! Joint stereo takes advantage of the way sounds distribute themselves over a normal stereo field. If you’re transmitting two signals that don’t comprise a normal stereo field — as would be the case if you’re transmitting two totally independent audio channels — it won’t save you anything.
Page 135 probably the most widely used system. In our view, this technology is acceptable for mono voice where high fidelity is not necessary. It is good also for cueing and intercom channels. We have included G.722 in ZephyrXstream because: • It had been the most popular coding method early on, so there are many of these codecs in use. Because it is a standard, codecs from various manufacturers have a good probability of being able to interwork with one‐another. (We’ve tested with many units and have found no problems so far.) • G.722 has the lowest delay of all popular coding methods. This method was invented in the late 70s and adopted as a standard in 1984 by the CCITT, the Consultive Committee for International Telephony and Telegraphy, a division of the United Nations. The technique used is Sub‐Band ADPCM (Adaptive Delta Pulse Code Modulation), which achieves data reduction by transmitting only the difference between successive samples. G.722 does this in two audio frequency sub‐bands: 50‐4kHz and 4kHz‐7kHz. DEEP TECH NOTE! Only two bits are allocated per sample for audio frequencies above 4 kHz – sufficient for conveying the sibilance in voice signals, but not very good for intricate musical sounds. Also, the “predictor model”...
Page 136 In cases where you are unable to get a good connection you may find this characteristic desirable, however you will find that the fidelity will be substantially less under these circumstances. CASCADING This section is preliminary, as coder cascading is an active field of investigation among algorithm designers, standards organizations, and users. Telos urges you to be wary and to let your own ears be the final judge until better information becomes available. Some of what we do know: • Some CCIR tests have demonstrated that one pass of Layer‐3 at 56/64kbps ...
Page 137 • Use the maximum bitrate you can afford at each stage. Hard disk recorders and other studio systems often have an option to adjust this. For very critical work, remember that Zephyr Xstream has several modes where a mono program is split over two digital network channels, thereby cutting the compression ratio by 50%. • Get the MPEG2 AAC and MPEG4 AAC‐LD advantage on low bitrate channels. • When using dedicated lines, consider using V.35 Single mode (see Appendix 6 ‐ Special Operational Modes) at 256 or 384 kbps to allow "headroom" for other coding in the chain. Mixed MPEG Signal Chains What about the case where you will be using Layers 2, 3 and AAC in a signal chain? It turns out that the two methods are nicely complimentary. At low bit‐rates, AAC or Layer‐3 gets more signal‐to‐mask margin than Layer‐2. This is why they perform better in the low bit‐rate tests. It accomplishes this by using a filter bank with more bands, 2048 vs 576 vs 32. One effect of this is “time spread.” (More frequency resolution requires a longer time window. It’s a law of physics thing...) For a small number of passes (one or two), this is good, as the ear has masking in the time domain as well as the frequency domain, and Layer‐3 naturally exploits this additional dimension. The down‐side is that when many stages of Layer‐3 are used at low bit‐rates, the time spread can become audible (softening of transients and pre‐echoes, mostly), and this is a bad thing. While Layer‐2 does not have this problem, it has another. Since it is closer to the edge for signal‐to‐noise, multiple generations result in unmasking (noise and grit, mostly). But the ISO/MPEG people do not propose that a bunch of passes of Layer‐3 be used. The idea is that AAC or Layer‐3 be used at ISDN/SW56 bit‐rates for field pick‐up and that Layer‐2 be used at higher bit‐rates in other parts of the signal path. This is why the ISO group decided to recommend the Layers as they did: AAC for 64kbps/channel and Layer‐2 for equal to or greater than 128kbps/mono channel. Our own experiments with codec cascading confirm that this is the right approach: the two coding methods seem to complement each other. Two passes of Layer‐3 (stereo @ 128 kbps) sound noticeably better than two of Layer‐2; a pass of Layer‐3 followed by a pass of Layer‐2 also sounds better than two of Layer‐2. Moreover, we’ve had customers who have used a pass, or two, of Layer‐3 followed by SEDAT without evident problems. USER’S MANUAL Section 6: AUDIO CODING REFERENCE 125...
Page 138: Choosing The Best Coding Mode For Your Application
Choosing the Best Coding Mode for Your Application The following chart summarizes and compares a number of important parameters for the coding modes offered in the Zephyr Xstream. Perceptual + Algorithm Type: Huffman + Temporal noise shaping + Backwards adaptive Linear prediction. Audio Freq Response (+0/-1 dB): Mono (56 or 64kbps) & Discrete Stereo (112 or 128kbps): Sample Rate: 16 kHz 24kHz 32 kHz...
Page 139 Delay : Stereo (128kbps): Sample Rate: 32 kHz 242 msec 48 kHz 190 msec Mono (64kbps): Sample Rate: 16 kHz 24kHz 32 kHz 284 msec 48 kHz 240 msec Stereo (64kbps): AAC @ 32kHz SR 404 msec 16:1 - 18:1 Coding "Power"...
Page 140 It should now be clear why we included AAC and AAC‐LD in the Xstream. These modes break the rules: AAC offers the best fidelity with delay less than Layer‐2, while AAC‐LD offers the shortest delay of any perceptual coder, while still offering near‐CD quality. HOT TIP! The "round trip" delay in a typical broadcast remote may be minimized by using the AAC-LD algorithm from the remote site to the studios with the return path using G.722. Even with a stereo feed to the station, the round-trip delay will be under 100 msec using this combination! This makes two-way conversations between the two sites much easier.
Page 141 DEEP TECH NOTE! While the "Stereo" modes keep the audio separate, the psychoacoustic model assumes the listener will be hearing both audio channels (i.e. it assumes that audio from one channel may mask audio on the other channel). For this reason the Mono/Dual mode is preferred when discrete unrelated material is to be sent down the two channels.
Page 142 Zephyr Xstream's L3 Dual/mono decoder allows for completely independent reception. Therefore, it can accommodate incoming L3 mono from two sites simultaneously without compromise. Of course both Zephyr Classic and Zephyr Xstream can transmit L3 mono audio to two sites. • A typical application which works perfectly with these modes is the one ...
Page 143 Connectivity The Xstream's MPEG bit streams follow standard ISO format, and may be used to communicate with any codec that supports this standard. In modes that require two ISDN channels, the channel‐splitting (IMUX) method becomes an issue. Layer 2 is widely supported by many manufacturers. The Zephyr Xstream supports the CDQ splitting scheme, so it may be used to communicate with the codecs from a variety of vendors that support this mode. In Layer‐3 Zephyr set the standard. Other codecs that support Layer‐3 use the FhG/Telos protocol developed for Zephyr classic (Buchta splitting). Telos has long been a proponent of international standards for codec interoperability. The specifications of our IMUX system, developed for AAC and AAC‐LD, is available without royalty, and is published on our web site for reference by other manufacturers. The Xstream’s G.722 mode offers connectivity with almost all codecs that use this coding method, and which do not use the rare H.221 framing scheme. We support the more common "SRT" (Statistical Recovery Timing) method. Ancillary data transmission is not standardized and is therefore not available between codecs made by different manufacturers. USER’S MANUAL Section 6: AUDIO CODING REFERENCE 131...
Page 144 USER’S MANUAL Section 6: AUDIO CODING REFERENCE 132...
Page 145: Isdn Basics
ISDN BASICS Background It was the introduction of digital transmission services from telephone providers, which has made the Zephyr family possible. The telephone infrastructure has moved from analog to digital. Telephony made the first significant use of digital audio techniques. In the mid 60’s a digital transmission method called “T‐carrier” began to be widely deployed to expand the voice‐ channel carrying capacity of existing copper wires. While they were intended originally for simple single‐channel analog, engineers discovered that the common copper wire pairs were capable of much higher bandwidth than the 3.4 kHz required for speech. Indeed, it was determined that two of these pairs could be made to relay 24 voice conversations – if they were digitized and appropriately multiplexed. Thus was born the basic technology used for digital telephony today. The standards developed then continue to define the digital telephone network: an 8 kHz sampling rate (resulting from the desired 4 kHz Nyquist frequency to accommodate a 3.4 kHz audio bandwidth, with guard band) with 8 bits of amplitude resolution (instantaneously companded to provide performance roughly the same as a 13 bit linear system producing 78 dB dynamic range for speech signals). Thus the basic voice channel bit rate was established to be 64kbps. (8kbyte/sec x 8bits = 64kbps.) DEEP TECH NOTE! Actually, the original Telco technique used only 7 bits for the audio, and the 8 for signaling. Since the digital technology was deployed as isolated links (later over microwave or coax cable) within an otherwise analog network, channel banks at the boundaries of each digital interface performed analog-to-digital-to-analog conversion.
Page 146: The Isdn Basic Rate Interface (Bri)
ISDN lines where the D channel is used only for signaling are sometimes referred to as 2B+0D rather than 2B+D service. D channel packet service is not widely available and is not used by Zephyr Xstream. ISDN BRIs are perfectly matched to Zephyr Xstream’s transmission capabilities. One channel ...
Page 147 DEEP TECH NOTE! The actual 2 B channels (i.e. timeslots) of the BRI are assigned on a per call basis. Therefore, from a theoretically correct viewpoint, Lines 1 and 2 do not fully correspond to B1 and B2. From the perspective of telephone network routing, each B channel appears to be a separate line with its own number and independent dial‐out capabilities. Since each has to be dialed or answered separately, they appear to be “lines” to users also. To reduce confusion (hopefully) ...
Page 148 In an emergency, to save a remote, you might be able to get a line to work without the proper SPIDs by dialing into it. If it works, this fix is only temporary, and the line will fail to initialize next time the Zephyr Xstream is booted. ISDN TIP! If you see SPID ERROR as your line status, your SPID for that channel is incorrect.
Page 149 ZEPHYR CLASSIC VS XSTREAM COMPARISON! Most versions of the Zephyr Classic can operate without Directory Numbers. Zephyr Xstream requires the DNs be present to receive incoming calls. Multiple Subscriber Numbers (MSNs) MSNs perform the same function when the ISDN protocol used is ETS 300 (Euro ISDN) or INS 64 (Japanese ISDN). In this case, use of the MSN is entirely optional. If no MSNs are entered, the first call to any of the numbers associated to the ISDN circuit will ring in as “line 1” if that “line” is idle. If “line 1” is in use the call will ring in on “line 2”. This is precisely the same behavior as when a line with the “AT&T Custom PTP” protocol is used. Long-Distance Digital Connectivity Long‐distance connectivity is routinely available in most parts of the USA from carriers such as AT&T, Sprint, and Verizon. Connectivity between any two given points is somewhat variable. The “dial 1+” default carrier may be chosen at the time you order the line, just as with traditional voice lines. Also, just as with voice lines, you may choose a carrier on a per‐call basis ...
Page 150: How To Order Isdn
DEEP TECH NOTE! The 56kbps limitation arises from a quirk of the older telephone infrastructure. The channel banks that have been widely employed in the long-distance network have a native 64kbps capability, but “rob” the low order PCM bit on every sixth frame in order to convey supervision information (on-hook/off-hook and dial pulses).
Page 151 CSD and CSV Recall that each ISDN BRI has two possible B channels. It is possible to order a line with one or both of the B channels enabled – and each may be enabled for voice and/or data use. Phone terminology for the class of service is “CSV” for Circuit Switched Voice and “CSD” for Circuit Switched Data. CSV is for calls using standard voice phone service and allows ISDN to interwork (i.e. to call) with analog phone lines and phones. CSD is required for Zephyr connections. Even though you may be sending voice, the codec bitstream output looks like computer data to the phone network. “Alternate CSD/CSV” or “Alternate voice/data” means both capabilities are supported. HOT TIP! While the D channel is always present for signaling purposes, it can, in theory, be used for packet data connections. ISDN lines where the D channel is used for signaling only are sometimes referred to as “2B+0D”...
Page 152 Zephyr Xstream ISDN Compatibility Protocols The Telco network and the Zephyr Xstream communicate via a ‘protocol’ – the language the user equipment and the telephone network use to converse (on the D channel) for setting up calls and the like. This is where there are differences depending upon the central office equipment used on the line and the standards which are supported by that particular equipment. While each type will work with the Zephyr Xstream, the differences need to be taken into account when lines are ordered and the Zephyr Xstream set up. In the USA, telephone companies use either AT&T/Lucent 5ESS, Northern Telecom/Nortel DMS‐ 100, or Siemens EWSD switches. Each of these can support the National ISDN 1 (NI‐1) protocol standard, which has been specified by Bellcore, the technical lab that was jointly owned by the phone companies. However, both AT&T and Northern Telecom had “custom” versions of ISDN which pre‐date the NI‐1 standard and a few switches have not been upgraded to the newer NI‐1 standard. CURIOSITY NOTE! There is also a newer National ISDN-2 (NI-2) standard, but it designed to be compatible with NI-1 for all of the basic functions.
Page 153 The Zephyr Xstream supports all of these with the appropriate selection of the "Telco" menu‐ selection in the TEL menu, as follows: TELCO SWITCH BRI CIRCUIT PROTOCOL AT&T/Lucent 5ESS Custom PTP (Point- to-Point) AT&T/Lucent 5ESS National ISDN-1 National ISDN-2 NORTEL DMS-100 Custom Functional NORTEL DMS-100 National ISDN-1 National ISDN-2 Siemens EWSD National ISDN-1 (USA) JAPAN NTT INS 64 EUROPE & Others ETS 300 (Various switches) Remember, complete information, as well as forms for ordering ISDN, are in Appendices 7 & 8 ...
Page 154: Switched 56
USER’S MANUAL Switched 56 Switched 56 is a transition technology, which is often available in areas of the USA & Canada where ISDN is not available. As the name suggests, this is a service that permits a single bi‐ directional digital data stream to be sent at a 56kbps rate. It can use either one or two copper pairs (“2 wire” and “4 wire”, respectively), depending upon the particular Central Office configuration. Be sure to determine in advance which service will be provided. Of course two Switched 56 circuits can be ordered, if transmission of stereo is desired. In most areas, switched 56 and ISDN are “inter‐worked” within the Central Office switches, so that subscribers of both services can connect to each other (at a 56kbps rate, of course). The Xstream can be used with Switched 56 and other synchronous data services through the optional V.35/X.21 interface and external CSU/DSU(s). We cover more about using the V.35/X.21 interface to interface to Switched 56 CSU/DSUs in Section 9 (The V.35/X.21 Interface). Section 7: ISDN BASICS...
Page 155: Livewire Tm - Ip Audio ............................................................................................................... 143
8 LIVEWIRE – IP Audio Configuration of the Zephyr Xstream for Livewire Overview Livewire can now be used to connect your Xstream to your studio’s Axia audio network. Livewire greatly simplifies wiring plus gives you all of the advantages of networked audio to and from your codec. The Xstream has been improved and updated many times since its original release. A Livewire‐ capable Xstream will have a version number that ends in 'b' or 'r' and it will have a 100Base‐T Ethernet jack on the back. Livewire has been included since the first release on that hardware. You will also notice that these Xstream codecs have grey “Axia‐style” web pages. Older Xstream codes many not be capable of Livewire due to hardware limitations. A non‐ Livewire‐capable Xstream will have a version number that ends in 'i' or 'p' and it will have a 10Base‐T jack on the back. The Zephyr Xstream has three types of local audio connections; analog, AES and Livewire audio (Note: Xstreams with a mixer do not have AES as an option). The audio configuration is established by the front‐panel menu of the Xstream. In the “Audio” menu, Input Source selection, you will find these three options. You would normally use Livewire at your studio and analog at the remote site. In addition to audio connections, you can use Livewire to manage GPIO at your studio location. GPO (output) control is available simultaneously at both rear panel parallel port and Livewire. GPI input mode must be specified as either Livewire or rear panel. If you are already familiar with Axia’s audio nodes, you may think of this new version of the Zephyr Xstream as an “original” Xstream codec with a 2x2 node on the front of it. The LAN port will be used for all Livewire audio connections in and out of the codec. As with all Livewire devices, the Livewire streams being delivered to the network require unique ID numbers. ...
Page 156 • If the stream being received by the codec is dual mono (such as G.722), the decoded audio is split between the two Livewire sources. The first source has the A mono feed replicated on both left and right channels. The second source has the B mono feed on both channels. Thus ‐ when you are using dual mono, you will use both codec outputs as independent Livewire sources. • Remember, Livewire streams are always stereo. In the case of a mono source as described above, the mono data is simply replicated on both left and right channels. • Dual/Mono TRANSMIT works only in Layer 3 Dual/Mono mode. In Layer 3 Dual/Mono transmit; the left channel of the first destination is used to create the “Channel A” mono feed to be sent over ISDN. The second destination is used for the “Channel B” mono transmit feed. The right channel of each of those streams is ignored. A Word about Livewire Codec Backfeeds Backfeeds to a codec in the Livewire channel of the backfeed is a L/R mono sum of the configured program source (usually PGM1). The left channel is the same L/R sum PLUS talkback. The left channel program can be dimmed or muted when talkback is present. Livewire Configuration Settings It is important to get the basics configured properly before you attempt to configure Livewire audio. Before we can get to the web pages for configuration of the Xstream, you will need to give the Xstream an IP address from the front panel controls. This process is well covered in Section 11.6 of the User’s Manual. Once you have assigned a unique IP address to the Xstream, it can be connected to your network. To access the web pages, simply launch your favorite web browser and “aim” it at the Xstream by entering the Xstream’s IP address in the browser’s URL box. Let’s now look at the Xstream’s Livewire configuration web pages in detail. Let’s start with the important items on the Administration and drill down to the other items in the sequence that you might use to setup your Xstream. ...
Page 157 Administration IP Settings This is where you will give the Xstream a name and configure the IP settings. The web page looks like this: Host name A 12 digit, alphanumeric name for this Node that may include hyphens but NOT spaces; those will be converted to hyphens. This name is used to identify the node on the network. You may wish to include the location of the node (studio or rack) in the name for ease of reference. In the example above, we have chosen a simple name of ZXS‐II. Network address (IP Address) This is the IP address of the Xstream codec. As you might expect, each Livewire device must have a unique IP address. In the example above, we have chosen an address in the 192.168.0.x range. If you are looking at this page, you have obviously already assigned an IP address to the unit and it is connected to the same LAN as your PC. Note that this IP address must be in the same range as your other Livewire devices and the Xstream must be physically connected to the Axia LAN. Otherwise, the Xstream will not be able to send or receive Livewire streams to/from your Axia network. USER’S MANUAL Section 8: LIVEWIRE – IP Audio...
Page 158: User Password
IMPORTANT! If you change this unit’s IP address, you will lose your browser connection when you click Apply. It will be necessary to reconnect using the new IP address. Gateway (Router) A gateway will be required if you are using the IP streaming method of communicating to another Xstream codec. NTP Server An NTP server is optional and not currently required for the Xstream. User password By default, the user name is “user” and the password is blank. We suggest you leave it this way until you get everything operating properly. Firmware version As with all other Axia devices, the Xstream has two internal memory banks. Each bank contains room for a complete version of operating software. This approach allows a software update download to be completed and checked without danger of making the unit inoperable if the download were to be incomplete or corrupted. It also provides an easy way to try a new software version and return to the old version if desired. The software version in each bank is displayed here. To change banks simply click in the “radio button” for the desired bank and then click on <Apply>. IMPORTANT! The node will reboot after you click Apply if you change the software version. This will result in loss of audio locally, and at any unit using the Xstream’s Livewire sources Saving Bank 1 Software New software is always downloaded to Bank 1 (the secondary bank). Downloading software (see below) will overwrite the software currently in this bank, if any. If you wish to save the software currently residing in Bank 1, you can save it by moving it to memory bank 0 by checking and applying the box designated with “commit this version to Bank 0”. Once you perform this USER’S MANUAL Section 8: LIVEWIRE –...
Page 159 USER’S MANUAL action, any software version that was previously located in Bank 0 will be replaced by the new, committed version. This action will not cause a reboot. Livewire QoS This configuration page provides the settings for Livewire QoS (Quality of Service). In almost all cases, the factory default settings are recommended so you will probably not need to make any changes to this page, particularly to the Audio Stream settings. It is worth noting here that the Livewire clock master priority will be set to either “0 (always slave)” or “3 (default)” and the Livewire clock mode will be “IP low rate (default)”. Detailed information on these settings is provided below. Here is the web page: Livewire Clock Master Since a device such as a codec is often powered‐down or relocated, the recommended clock priority setting for an Xstream is “0 (always slave)”. Here is a bit of background on this topic. Livewire’s clocking system is automatic and largely transparent to end users. By default the Axia hardware node with the lowest Ethernet MAC address will be the clock “master”. The system will automatically and transparently switch to a new unit as clock master if need be. We do, however, permit a forced clock master to be assigned to a particular device (usually a node), or to set certain nodes as “preferred” for clock mastership while maintaining automatic operation. For example you may wish to assign nodes that are on UPS power as preferred clock masters. Note that in the automatic modes, a new clock master is designated only when the current master becomes unavailable (adding a new node will not change clock mastership regardless of the new node’s setting). The only exception to this rule is the 7 (Always Master) setting. Section 8: LIVEWIRE – IP Audio...
Page 160 The following table provides the settings possible for choices Live Clock Priority and provides some information on each of these settings: 0 (always slave) “STL” Unit will never be master and is only used with Ethernet radios. 0 (always slave) This unit will never be used as clock master. 3 (default) The usual and default setting for an 8x8 node. Nodes with this setting will be used as clock masters before 4 (Secondary Master) those set to 3. Nodes set to this setting will be used as clock masters before 5 (Primary Master) those set to 4. This forces a particular node to be clock master, even if another 7 (Always Master) node is currently clock master. If this mode becomes available then the usual prioritization is used. This setting forces a particular node to be clock master. Use 7 (Always Master) “STL only when two nodes are connected back to back without an Snake” Ethernet Switch. IMPORTANT! Only a single node on a Livewire network should ever be set to 7 (Always Master). For this reason we do not recommend using that selection. Livewire Clock Mode Provided for compatibility with older revisions: IP low rate (default) – recommended setting Ethernet – compatible with 1.x firmware IP High rate – compatible with 2.1.x master Receive Buffer Size ...
Page 161: Livewire Audio Settings
USER’S MANUAL 801.1p tagging, 802.1p VLAN ID, 802.1q Priority, & DSCP Class of Service 802.1p tagging is necessary within the Livewire network to mark high‐priority audio packets. This information is used by the Ethernet switches in the packet scheduling and queuing mechanism. It provides low‐jitter packet forwarding of Livewire clock and low‐latency audio streams. On the other hand, Standard streams don’t need tagging, because they are not low‐latency. By default, standard streams are marked with Type of Service (DSCP code points) information in the IP header which can be used by L3 switches to provide better service to our audio streams than to best effort IP traffic. There is an option to enable L2 802.1p tagging on standard streams, and this may be used with switches which do not use the DSCP information included in the TOS field of the IP header. We do not enable this tagging by default, because it wouldn’t work in cross‐over Ethernet connection to PCs; most network cards do not accept 802.1p frames by default. You should not need to change these default settings unless you are building a system which is not based on our recommendations. In Axia nodes and the Xstream, the VLAN ID setting is a read‐only value. It is always 0 and cannot be changed. As a result Livewire audio always uses the native VLAN assigned to the port of the switch. “DSCP Class of Service” is a standard describing the tagging of IP frames with service information. Network equipment can be set up to provide different forwarding delay and drop precedence depending on the service information. Our defaults are compatible with most Ethernet equipment defaults for class of service Livewire requires; you should not need to change them unless instructed by Axia Support. Livewire Audio Settings There are two ways to setup Livewire audio, Basic Mode (shown by default) and Expert Mode. The difference between these two modes is that Expert Mode allows you to customize the codec backfeeds. We will take a look at both scenarios. Livewire Configuration – Basic Mode This mode will be fully adequate for most users. It assumes the default backfeed for the codec. This is the automatic mix‐minus created by the mix engine and will consist of the PGM or AUX bus as configured on your Element or other control surface. ...
Page 162 USER’S MANUAL excellent” audio. You have a channel number range available from 1 to 32,767 so there are lots of numbers to choose from. In Basic Mode, the backfeeds will have the same channel number as the source channels and will automatically be configured as “to source”. This ensures that your backfeed consists of a suitable mix‐minus that is automatically created by your Studio Mix Engine. Livewire Configuration – Expert Mode There may be a case when you want to use something other than the automatic mix‐minus as a backfeed. For example, maybe you want to use an AUX bus to create your own custom mix. In this case, you would use Expert Mode to create the setup. Use the destination drop‐down boxes to select the Livewire streams that you wish to use as backfeeds to your codec channels. Remember that these destinations are used to designate the audio being delivered to the codec inputs. Select “from source” if the Livewire channel you are selecting is not one of the automatic backfeed channels. Click “apply” and the selected streams will be delivered to the codec channel inputs. In this example, the channels used as backfeeds illustrate the “normal” settings you would get when Basic Mode is used. Changes can be made to these basic settings as required and applied. Section 8: LIVEWIRE – IP Audio...
Page 163 USER’S MANUAL System Settings (GPIO) This page of the Xstream configuration has the GPIO input mode setting. Since Livewire has the capability of “embedded” GPIO, there is often no need for connections to switches as a means of controlling various external functions. You can use Livewire or the traditional parallel port to manage GPIO at your studio location. GPO (output) control is available simultaneously at both rear panel parallel port and Livewire. GPO is sent to both the back panel and Livewire, and is triggered by the events specified on the System menu. GPI (input) mode must be specified as either Livewire or rear panel since we can take inputs from the rear panel OR Livewire, not both at the same time. GPI comes from either Livewire or the back panel, and triggers the event specified on the front panel Sys menu (or System web‐ page). This can be either a dial setup or sent through ancillary data to the remote end. GPIO implementation can be tricky. Please check our web site for new application notes on this topic and other topics that may involve complex configurations. Section 8: LIVEWIRE – IP Audio...
Page 164 USER’S MANUAL Section 8: LIVEWIRE – IP Audio...
Page 165: The V.35/X.21 Interface Option
IMPORTANT! Always disconnect AC (mains) power and Telephone Circuit Before removing the Zephyr Xstream’s cover. Potentially lethal voltages are inside! Failure to observe the above warning may result in injury or even death! Refer all servicing to qualified personnel.
Page 166: Uses For The X.21/V.35 Interface Option
6. Once the connectors are aligned, push the V.35/X.21 option card straight down into the connector on the motherboard. Only moderate pressure should be required to seat the card. 7. Fasten the option card to the rear panel of the chassis using the two 4‐40 button head screws previously removed in step 3, above. 8. Replace the lid. If you have a Zephyr Xstream MXP, align the blue bumpers with their corresponding holes. 9. Fasten the lid using the 18 (or 14) screws removed in step 1. 10. To use your V.35/X.21 interface you will need to connect an appropriate cable: • For V.35 use Telos part # 2091‐00023 (old #9812‐0000) • For X.21 use Telos part # 2091‐00024 (old #9822‐0000) 11. To enable the X.21 Interface, change the Interface menu selection In the TEL menu to V.35 Split or V.35 Single as appropriate for your application. Uses for the X.21/V.35 Interface Option SWITCHED 56 Switched 56 is a transition technology which is being offered primarily in the few areas of the USA & Canada where ISDN is not available. As the name suggests, this is a service that permits a single bi‐directional digital data stream to be sent at a 56kbps rate. A switched 56 line requires a Channel Service Unit/Data Service Unit (CSU/DSU) to interface the digital line with the Zephyr. V.35/ X.21 INTERFACE (MINI-40 PIN) Zep hyr™...
Page 167 Adtran 901 Explorer Ave Huntsville, AL 35806 800‐827‐0807 www.adtran.com Adtran makes all sort of equipment for the telecom industry. The Telcos buy equipment from Adtran. Reputable company. Most USA Telos Dealers carry Adtran equipment. An appropriate CSU/DSU (Channel Service Unit/Data Service Unit) is required to interconnect the line with the Zephyr Xstream. Dialing is performed using the CSU/DSU directly. Popular models are the Adtran DSU III S4W (for 4 wire circuits, Adtran part # 1202013L1) and DSU III S2W (for 2 wire circuits, Adtran part # 1200015L2) see www.adtran.com or your Adtran dealer. Adtran also makes CSU/DSU's for use with dedicated 56 or 64kbps circuits. See below. If the location with Switched 56 is only to be used temporarily, you might wish to rent the CSU/DSU rather than purchase one. One rental source is: Silver Lake Audio 90 South Long Beach Rd Rockville Centre, NY 11570 516‐763‐1776 www.silverlakeaudio.com In most areas, switched 56 and ISDN are “inter‐worked” within the Central Office switches, so that subscribers of both services can connect to each other (at a 56kbps rate, of course). Therefore, the far end unit may be on ISDN or Switched 56. If two Switched 56 circuits are available (as well as two CSU/DSU's) Stereo or Mono 128 operation is possible. T1 Circuits This permits subscriber access to up to 24 channels of 64kbps each. T1 circuits into the dial up network are frequently used with private telephone systems (i.e. PBXs). However, dedicated T1 service between two locations is available. This option has become increasingly popular for STLs. T1 requires a special Multiplexer or DSU (available from Adtran and others) to break out the individual channels. These are usually modular, permitting each channel (time slot) to be used individually as desired. Each Zephyr Xstream would require 1 or 2 channels for bi‐directional mono and stereo, respectively, if the V.35 split mode were used. In that case, each channel would be broken out by the multiplexer into a separate V.35/X.21 port. Or a single 112, 128, 256, 0r 384 kbps connection can be used, if the Xstream V.35 single is selected for the Interface in the TEL menu. See Appendix 6 (Special Operational Modes) for ...
Page 168 We recommend that option for STLs and other critical audio paths. Equipment is available from a variety of vendors to allow uncompressed audio transmission over T1 circuits. Zephyr Xstream's V.35 Single offers audio connectivity at an intermediate cost while offering the high fidelity and robustness possible with MPEG at low compression ratios.
Page 169: Hook-Ups To Non-Isdn Synchronous Networks
V.35 is a standard for connecting to synchronous digital data paths. Like RS‐232, it defines signals and (not officially) connectors and pin‐outs so that equipment from various manufacturers may talk with each other. The usual connector is a big boxy AMP type, which was chosen by AT&T decades ago. Most terminal equipment sold for the US market supports the V.35 standard (although an adapter cable may be required to connect to the usual connector). The following Signals are provided at the V.35 end of the Telos cable: V.35 CONNECTOR PIN-OUT Description Ground RTS (Request to Send) CD (Carrier Detect) DTR (Data Terminal Ready) TX Data...
Page 170 RS-232). The transmit voltage level is ± 5.5V, but receivers must handle wider swings. X.21, a more modern standard, is also used for connections to synchronous digital data paths. This standard specifies the much more reasonable DB‐15 connector. Fortunately, it is possible to design an interface that accepts both V.35 and X.21 signals. This we have done with the Zephyr Xstream, allowing connection to both types just by changing the interconnect cables. The X.21 standard is frequently seen in Europe and other areas outside the USA & Canada. The following Signals are provided at the X.21 end of the Telos cable: X.21 CONNECTOR PIN-OUT Description TX Data RX Data CD (Carrier Detect) Clock Ground...
Page 171 # 2091-00024 You can tell whether you have V.35 or X.21 by looking at the connectors. These are the “footprints” of the two types to help you select the proper Telos cable to interface to your datacom gear. Each V.35/X.21 connection conveys both send and receive digital signals. The V.35 or X.21 cable from Telos includes two V.35 or X.21 plugs. This two‐port cable may be ...
Page 172 We had to use a small socket in order to make more room on the rear panel (sorry), so you cannot plug directly into it with off-the-shelf cables. You must have the appropriate cable from Telos to connect! The Xstream uses the same cables as the Zephyr.
Page 173 V.35/ X.21 INTERFACE (MINI-40 PIN) Zep hyr™ Xstrea m Single port operation supporting 112, 128, 256, or 384 kbps for operation on a single V.35/X.21 port. The Interface Selection in the TEL menu should be set to" V.35 Single." The second port can be used to send an identical audio stream to a second site if desired.
Page 174 ZEPHYR CLASSIC VS XSTREAM COMPARISON Since the Zephyr does not have the L3 Dual/Mono Receive mode, data corruptions on either data channel will cause dropouts on both audio channels. Therefore, Zephyr users performing troubleshooting tests would need to switch to mono operation, over a single data channel, to verify each channel separately.
Page 175 In the direction CSU/DSU ‐> Zephyr Xstream, only CD (Carrier Detect))is implemented. This signal is used to tell the Zephyr Xstream which port/channel has been connected and activated. It is used for internal selection of the network clock source. When the TEL menu selection Interface is set to “V.35” the TEL status screen will indicate “Inactive” or “Active” depending on whether the CD signal is present for that channel. NOTE that at any given time only one of the V.35 signals is providing clock which is used for both ports. Therefore both V.35 sources must be at the same clock frequency (which is generally the case since Telco’s use highly accurate clock sources for all of their gear). Configuring CSU/DSUs There will generally be at least two main configuration menus. One for interface to the DTE (Data Terminal Equipment) and one for the interface to the “loop” or “line”. Your CSU/DSU manual (or manufacturer) should have information on the correct settings for the various options. DTE Options: Set to “Synchronous” and set the bit rate to match the Zephyr Xstream’s and line’s bit rate (usually 56kbps). If necessary, you may disable any handshaking signals by setting them to “forced” or “ignored” since handshaking is not required with synchronous data. “Anti streaming” or “scrambling” options should be disabled. Loop or Line Options: Set to the speed of your line (usually either 56 or 64kbps). The Xstream’s CODEC menu option Bit Rate must be set to the same rate as the line. Your Telco or CSU/DSU manual should be able to give you the correct setting for any other options. Clock Options: It is essential that the clock option be set to “network” or “from loop”. The Xstream will clock its synchronous data to the clock from the CSU/DSU. In order to prevent buffer overflow or underflow, the far end unit must share the same clock. The only way to ensure this, is for each CSU/DSU to derive its clock from the network (i.e. from the line itself). The CSU/DSU must not derive clock from the V.35 port. DEEP TECH NOTE! In cases where "dry line" modems are used one end can use a local clock (master) and the other end (slave) can derive timing from the far end by using loop timing.
Page 176 Channel Assignment: DSU's for use with T1 or E1 will have a menu that determines which channel(s) should be brought out (or aggregated and brought out) to each V.35 port. This will need to be set on each end. If you are using a fractiona1 T1 or E1 you will need to know which channels are active. The Telco can give you that information. USER’S MANUAL Section 9: The V.35/X.21 Interface Option...
Page 177: Applications Guide - Hints & Ideas
10 APPLICATIONS GUIDE – Hints & Ideas Of course we hope that you will (eventually) read through the other sections of this manual – we give you a lot of information about audio coding technology and lots of information on how and when to use the different coding modes. However, here we have included the following informational guide to get you started. 10.1 Xstream Versatility – types of dial-up connection and modes The Xstream is the most versatile codec platform on the market at this time, communicating with a variety of devices in several ways. In this section, we'll give a review of the different modes available when on ISDN, illustrating the various capabilities of the unit. First, let us look at a matrix of the Xstream's abilities: DIAL MODE TALKS TO NETWORK CALL TYPE Most ISDN Circuit ZEPHYR Codecs Switched Data (CSD) Zephyr Xport Circuit XPORT...
Page 178 USER’S MANUAL Xstream ISDN to ISDN Connection Examples: Mono Mono ISDN ISDN Telephone Network (1 Call) (1 Call) Mono Mono The basic Mono 56 or 64 kbps ISDN codec connection is shown above. The audio quality is dependent on the settings in the CODEC menu. ISDN is used end‐to‐end for the best quality and lowest delay. When using MONO modes the channel A inputs and Line 1 are used. The dialing MODE for this call is Zephyr. Mono Mono ISDN ISDN Telephone Network (1 Call) (1 Call) Mono Mono The call between an Xport on ISDN and the Xstream is similar to the above case. In this case the Xstream would be set to AAC‐LD Mono‐64 @ 48kHz. No configuration of the Xport is necessary. The dialing MODE for this call is Zephyr. ...
Page 179 Mono ISDN (1 Call) Mono Mono ISDN (1 Call) Mono Dual mode operation allows a "central" codec to connect to two "remote" codecs simultaneously. Dual operation is available in both Layer 3 and G.722. The remote codecs generally would both use the A inputs and outputs and Line 1. The central codec will use the A inputs and outputs to communicate with the remote site connected to Line 1, and the B inputs and outputs to communicate with the remote unit connected to Line 2. The dialing MODE for these calls is Zephyr. Mono ISDN (1 Call) Mono Mono ISDN (1 Call) Mono This example is the same as the above, only the connection shown is with Xports as the field. In this case, G.722 would be used. The dialing MODE for these calls is Zephyr. USER’S MANUAL ISDN Telephone Network (2 Calls) 2 X Mono ISDN Telephone Network...
Page 180 Mono ISDN (2 Calls) Mono The basic Mono 112 or 128 kbps ISDN codec connection is shown above. The audio quality is dependent on the settings in the CODEC menu. ISDN is used end‐to‐end for the best quality and lowest delay. In Mono 128 modes, the channel A input is used. The dialing MODE for these calls is Zephyr. Mono ISDN (1 Call) Stereo Mono ISDN (1 Call) Stereo The Stereo 64 mode is a special case. In this scenario a central codecs sends the same stereo audio to two sites. The remote‐to‐central return path is mono and can be L3 or G.722. The dialing MODE for these calls is Zephyr. USER’S MANUAL Mono ISDN Telephone Network (2 Calls) Stereo ISDN Telephone Network (2 Calls) Section 10: APPLICATIONS GUIDE Mono 2 X Mono...
Page 181 Xstream POTS to ISDN Connection Examples Mono POTS (1 Call) Communications Grade Mono The "integrated services" of ISDN means ISDN line can communicate with POTS lines. This arrangement means that the connection is mostly digital, with only the POTS line analog, thereby giving better performance. The dialing MODE for this call is Xport. No other configuration is required in this case. POTS (1 Call) The Xstream can also act as a digital telephone coupler. The dialing MODE for this call is Phone. No other configuration is required in this case. USER’S MANUAL ISDN Telephone Network (1 Call) 2 X Mono ISDN Telephone Network (2 Calls) Section 10: APPLICATIONS GUIDE Mono Broadcast Quality...
Page 182: Choosing The Best Transmit Mode For Your Application
AAC should be used whenever it is supported at both ends. Offers 15 or 20 kHz audio frequency response, at 32 or 48 kHz sample rate, respectively COMPATIBILITY TIP! Zephyr Xstream supports both 32kHz and 48kHz sample rate for Layer-2 in the following modes. The majority of Layer-2 codecs deployed in the field (including the Zephyr Classic) support the 48 kHz sample rate only. The Layer 2 Mono/Half mode allows operation at 24 kHz sample rate, however the delay is significantly increased.
Page 183 Rate selected in the CODEC menu. Zephyr VS Zephyr Xstream ! Zephyr does not offer Layer 3 Mono 128 operation (although the Joint-stereo mode can be used as a substitute for it). However, the Zephyr can decode the Xstream's Layer 3 Mono-128 streams. Set the Zephyr to L3 Stereo Rcv.
Page 184: Applications Guide
USER’S MANUAL Also note that Layer‐3 has more delay than Layer‐2 which has more delay than AAC. AAC‐LD follows with G.722 having the shortest delay. Note that for a given bit rate and sample rate, that AAC offers the best fidelity, followed by Layer‐3 and AAC‐LD, followed Layer‐2, and finally by G.722. As a Zephyr Xstream user, you have the best of all worlds. Layer‐2, Layer‐3, and G.722 offer backwards compatibility with older codecs, while AAC and AAC‐LD allow better quality or reduced delay. 10.3 Applications Guide Voice-over applications on a single B channel (1 call) Good: Use L3 Dual/Mono Transmit ‐ Far end would use L3 Mono (Zephyr) or L3 Dual/Mono (Zephyr Xstream) Receive. Use 32kHz sample rate for best fidelity. Best Quality: Use AAC Mono 64 transmit ‐ Far end should use AAC 64 for Receive Voice-over applications using two B channels (2 calls) Good: Use L2 Mono 128 Transmit – Far end should use L2 Receive (Zephyr or Zephyr Xstream) at 48kHz sample rate. Better: Use L3 Mono 128 Transmit – Far end should use L3 (Zephyr Xstream) or L3 Stereo (Zephyr) ...
Page 185 Voice-over applications communicating with non-Telos codecs Most Common: G.722 is the most commonly available coding mode in the world. Use G.722 for your Transmit & Receive modes. Note: Since the quality of G.722 is only moderate, you should only use this mode with clients who do not have a better option available. One B channel (1 call) required for mono. Most Popular: MPEG Layer‐2 Mono 128 seems to be the defacto standard used in the voice Industry. Set your Transmit to L2 Mono 128 and your Receive to L2. Two B channels (2 calls) required. Use 48 kHz sample rate Radio remotes (Mono) using a single B channel (1 call) Good: Field unit: L3 Dual/Mono Transmit and G.722 Receive Studio unit: G.722 Transmit and L3 Dual/Mono Receive (Zephyr Xstream) or L3 Mono (Zephyr) Use 32 kHz sample rate for best fidelity, or 48kHz sample rate for reduced delay Better: Field unit: AAC‐LD Mono 64 Transmit and G.722 Receive Studio unit: G.722 Transmit and AAC‐LD Receive This option has low round‐trip delay with fidelity comparable to Layer‐3 Use 32 kHz sample rate for best fidelity or 48 kHz for reduced delay. Best: Field Unit: AAC Mono 64 Transmit and G.722 Receive. Studio Unit: G.722 Transmit and AAC 64 Receive This option has better fidelity than the AAC‐LD, but longer delay. ...
Page 186 When full fidelity is needed in both directions: Field Unit: AAC‐LD Mono 64 Transmit and AAC 64 Receive. Studio Unit: AAC‐LD Mono 64 Transmit and AAC 64 Receive Use 32 kHz sample rate for best fidelity or 48 kHz for reduced delay Radio remotes where only POTS is available at the remote location Best: Field Unit: Zephyr Xport Studio Unit: Zephyr Xstream Dial using "Xport" mode Radio remotes (Stereo) using a single B channel (1 call) Good: Field Unit: AAC J Stereo 64 Transmit and G.722 Receive. Studio Unit: G.722 Transmit and AAC 64 Receive. Use 32 kHz sample rate. When best stereo is needed in both directions: Field Unit: AAC J Stereo 64 Transmit and AAC 64 Receive. ...
Page 187 • The same stereo audio will be received at both studios. Return audio from the studio connected to "Line 1" will be output "Channel A/Left" and Return audio from the studio connected to "Line 2" will be output "Channel B/Right". • L3 Dual/Mono can be used in place of G.722 if better fidelity from the studio to remote direction is required. Radio remotes (Stereo) using two B channels (2 calls) Good: Field Unit: Use L3 J Stereo Transmit and G.722 Receive (Zephyr or Zephyr Xstream). Studio Unit: Use G.722 Transmit and L3 Stereo (Zephyr) or L3 (Zephyr Xstream) Receive. Use 32 kHz sample rate for best fidelity or 48 KHz sample rate for reduced delay Better: Field Unit: Use AAC‐LD J Stereo Transmit and G.722 Receive. Studio Unit: Use G.722 Transmit and AAC Receive. Use 32 kHz sample rate for best fidelity. Best: Field Unit: Use AAC J Stereo Transmit and G.722 Receive. Studio Unit: Use G.722 Transmit and AAC Receive. Program Link Applications (Stereo) For bit rates of 112-128 kbps: For AM and FM backup Studio‐Transmitter links: Use V.35 Split or V.35 Single depending on whether two 56 (64) kbps channels or a single 112 (128) kbps channel is to be used. Studio Unit: Use AAC J Stereo Transmit. ...
Page 188: Multi Codec Connections
Transmitter site Unit: AAC or L2 Receive Use 48 kHz sample rate 10.4 Multi Codec connections Sometimes one Xstream will need to connect to two other units simultaneously. The Xstream's L3 Dual/Mono Receive mode allows for considerable flexibility in this regard. In all of the following cases the Bit Rate and Sample Rate must be matched on all three units. A single Xstream can receive Layer 3 mono from two sites as shown below: Xstream C Receive = L3 Dual/Mono A Zephyr Xstream set to L3 Dual/Mono can receive one channel from each of two remote Zephyrs. Each remote unit would be set to L3 Dual/Mono Transmit. If Xstream A is dialed from "Line 1" to Xstream C's "Line 1," then Xstream A's channel A audio would be present at Xstream C's channel A output. If Xstream B dialed from "Line 1" to Xstream C's "Line 2" then Xstream B's channel A audio would be present at Xstream C's channel B output. Return feeds to A and B can be made using G.722 or L3 Dual/Mono. Transmitting mono to two sites is also easily accomplished. In the above scenario Xstream C would use L3 Dual/Mono for Transmit. Xstreams A and B would be set to L3 Dual/Mono receive. If Xstream A were connected to "Line 1" of Xstream C, it would receive the channel A audio from C. If Xstream B were connected to "Line 2" of Xstream C, it would receive the channel B audio from C. USER’S MANUAL Xstream A Transmit = L3 Dual/Mono Xstream B Transmit = L3 Dual/Mono Section 10: APPLICATIONS GUIDE...
Page 189: Choosing A Receive Mode
"inverse multiplex" (IMUX) scheme must be used to break the standard MPEG data stream into two streams for transmission. In Layer-2, the Zephyr Xstream uses the CCS CDQ (TM) IMUX protocol. Most Layer-2 codecs support this method, allowing for compatibility among different codec manufacturers.
Page 190 HOT TIP! The Sample Rate and Bit Rate must be matched for the auto mode to detect the incoming bit stream. Certain conventions are used which it will help to understand: 64 ‐This is a Zephyr Xstream Transmit mode that requires only a single B channel to operate. The actual bit rate of that channel could be either 56 or 64 kbps. 128 ‐ This is a Zephyr Xstream Transmit mode that requires two B channels to operate. The actual bit rate of those channels could be either 56 kbps (for a total of 112 kbps) or 64 kbps (for a total of 128 kbps) Dual/Mono ‐ This is simply a special case of Mono 64. In the Dual/Mono case the A audio source is sent over "Line 1" while the B audio source is sent over "Line 2". J‐Stereo ‐ This is "Joint Stereo" and is the preferred stereo mode in most cases. The biggest exception is when stereo material will later be transmitted over a surround sound transmission system (i.e. for Video or Film production) in which case the Stereo mode should be used. All stereo modes require 2 B channels (2 calls) unless otherwise stated. J‐Stereo 64 ‐ This mode, unique to AAC, is the only stereo transmission mode that does not require 2 B channels. L2 Half 64 ‐ This is a special Layer‐2 mode that operates at 24 kHz sample rate. The audio will be transmitted at 24 kHz sample rate regardless of the actual sample rate selected in the CODEC menu. The following pages show the correct Xstream “receive” settings for various codecs: USER’S MANUAL Section 10: APPLICATIONS GUIDE...
Page 191 Zephyr Xstream at far end: Transmit Mode at Far End AAC-LD Stereo AAC-LD J-Stereo AAC-LD Mono 128 AAC-LD Mono 64 AAC J-Stereo 64 AAC J-Stereo AAC Stereo AAC Mono 128 AAC Mono 64 G.722 L2 Stereo L2 J-Stereo L2 Mono 64...
Page 192 Zephyr (classic) at far end: Transmit Mode at Far End: ° L3 DUAL or L3 MONO L3 STEREO L3 JSTEREO G.722 L2 MONO L2 MONO128 ° L2 STEREO or L2 DUAL L2 JSTEREO L2 HALF/24 Sample rates on the two ends must match * Actual Sample Rate will be 16 kHz regardless of setting + Actual Sample Rate will be 24 kHz regardless of setting °...
Page 193 Zephyr Xport at the far end When Xport is on ISDN use the "Zephyr" Dial Mode selection on the dialing Xstream or Xport: • Transmit ‐ AAC‐LD Mono 64 • Receive ‐ AAC‐LD Mono 64 • Sample Rate : 48 kHz • Connect on one B channel only When Xport is on POTS use the "Xport" Dial Mode selection on the dialing Xstream or Xport: • No configuration necessary Comrex Nexus at far end Use the following settings on the Xstream: • Transmit ‐ G.722 • Receive ‐ G.722 • Sample Rate ‐ Any • Connect on one B channel only. USER’S MANUAL Section 10: APPLICATIONS GUIDE...
Page 194 CCS CDQ Prima at far end Far End Setting Set Local Receive Mode (ID #): 6: G.722_56K:QS G.722 7: G.722_64K:QS G.722 24:ZEPHYR/56K:QS L2 64 25:ZEPHYR/64K:QS L2 64 26:ZEPHY/112K:QS 27:ZEPHY/128K:QS 30:LYR3IND56:QS L3 Dual/Mono 31:LYR3IND64:QS L3 Dual/Mono 32:LYR3IND56/32:QS L3 Dual/Mono 33:LYR3IND64/32:QS L3 Dual/Mono 36:ZEPHY112MONO:QS 37:ZEPHY128MONO:QS (2) = Both B channels must be dialed for this mode...
Page 195 CCS Roadrunner at far end Far End Setting Set Local Receive Mode (ID #): 1: L2/56/24YES:QS L2 64 3: L2/56/32YES:QS L2 64 5: L2/56/48YES:QS L2 64 7: L2/64/24YES:QS L2 64 9: L2/64/32YES:QS L2 64 11: L2/64/48YES:QS L2 64 13: L2112/32YES:QS 15:L2112/48YES:QS 17:L2128/32YES:QS 19:L2128/48YES:QS...
Page 196: Dealing With Delay
10.6 Dealing With Delay Mix-Minus All perceptual coders have too much delay for talent on remote to hear themselves via a round‐ trip loop without an "echo" that can he annoying and disorienting. Therefore, a special mix‐ minus (sometimes called a "clear feed" or "mix‐minus one") arrangement is required – exactly the same as has been used with satellite linked remotes for years. The European term “mix minus one” is clearer in the most typical situation. The principle is this: The remote talent does not hear himself via the studio cue return. Rather, his microphone is mixed locally with a studio feed that has everything but the remote audio – thus the “mix‐minus” designation. The announcer gets in his headphones a non‐delayed version of himself and a slightly delayed version of all of the studio pieces. Most modern digital console can be configured to generate mix‐minuses without user intervention. If you are using an older console that does not have this ability, or a non‐broadcast console, here are some methods you can use. Examples of mix-minus set-ups: Example 1: Using a broadcast console (mixing desk) The simplest form of mix‐minus involves using a dedicated bus to feed the Xstream. In this case, we show Audition feeding the Zephyr Xstream. Note that Fader 5 (a stereo input) is NOT assigned to Audition. USER’S MANUAL AUDITION PROGRAM Section 10: APPLICATIONS GUIDE...
Page 197 NOTE: This must be turned down or AUX 2 will no longer be a mix-Minus AUX 1 AUX 2 Mic1 Z XS A Example 2: Using a Utility Mixer The simplest form of mix‐minus using a Utility Mixer with mono inputs. Two mono faders and two busses, AUX 1 and AUX 2 are used. It is also possible to use the mixer's main outputs to form a mix‐minus, by Panning one Input channel "hard left" and the other input" hard right". Phones and Remotes To save money and hassle, callers are usually received at the studio, rather than at the remote site. In this situation, phones need to be fed to the remote talent, so that they can hear and respond to callers. Moreover, the phone callers need to hear the talent. In many cases, the remotes are sufficiently distant that the station cannot be monitored for the caller feed. Even if it could, the profanity delay would be a problem, since the talent needs to hear the phone pre‐ delay. The talent hears callers via the return path. As before, this return is fed with mix‐minus: a mix of everything on the program bus minus the remote audio. As for the second half of the equation, the callers hear the talent because the remote feed is added to the telephone mix‐minus buss. No problem if you have a set‐up that permits selective ...
Page 198 Remember, if any of the announcer audio from the remote site is returned via the monitor feed, it will be delayed by the codec link, causing an echo effect. Problem. The answer is to make sure you have the best possible hybrid with the maximum trans‐hybrid loss, such as a system that uses ISDN lines. If it has variable override (caller ducking), you could increase the amount when these remotes are in progress (or, on some hybrids, you would decrease the setting of the "duplex" control). Some users place a downward expander, carefully adjusted to gate the leakage, in the caller audio path. REMOTE SITE Diagram showing system set‐up for remotes with delay in the transmission path and phones taken at the studio. Note that this is the same as required for satellite links. Another issue worth considering is the round trip delay. The apparent on‐air response time of the talent to callers’ comments will be the sum of studio‐to‐remote delay + remote‐to‐studio delay + talent’s thinking time. For this reason the studio‐to‐remote path will generally use the G.722 mode, which sacrifices fidelity for delay (after all, the callers need not be in high fidelity). This round trip delay issue will also affect your choice of remote‐to‐studio coding. If the show will be 90% talk with just a small amount of music then delay can be minimized by using a mono mode for this path rather than using one of the slower stereo modes. Here is where Zephyr Xstream’s AAC‐LD encoding really shines. It gives you high quality announcer audio with the lowest delay possible. In addition, the reduced delay time will serve make any leakage less distracting (for more on this see Section 6 ‐ Audio Coding Reference). Another trick is to use a POTS call (call mode = Phone) for the studio‐to‐remote link, which will make the delay in that direction very small. Other intermediate tradeoffs are possible, and will be dependent on your format. Talent thinking time can be significantly reduced by drinking a strong cup of coffee! For information on the tradeoff between audio quality and delay, refer to manual Section 6 (Audio Coding Reference) USER’S MANUAL Other Sources Zephyr for music, etc. Trans- mission Path Studio Console...
Page 199: The Works! - Detailed Menu Reference
11 THE WORKS! - Detailed Menu Reference In this section, we cover each selection of every menu in detail. It is intended as a reference, and for those using some of the more specialized features of the Zephyr Xstream. Beginning users are likely to find the topics of interest covered in Section 4 (Basic Operation & Installation) and Section 4 (Guided Tour of the Menus) IMPORTANT! This section covers the front panel menus of the Xstream. Many of these functions are also available from the HTTP interface that can be accessed by using your favorite web browser. Note that you will need to have Java installed to view certain items such as the Livewire audio meters.
Page 200 MENUS - The CODEC, TEL & SYSTEM Menus CODEC CO DEC CODEC Compat. Zephyr CODEC S IP Port UDP Port HTTP Port TCP Port Section 11: THE WORKS – Detailed Menu Reference USER’S MANUAL (all models) SYSTEM SYSTEM SYSTEM SYSTEM 5060 9150...
Page 201 MENUS - The DIAL & AUTO Menus (all models) MANUAL DIALING DIAL Phon e # DIAL DIA L Phon e # DIAL DROP F our Times Section 11: THE WORKS – Detailed Menu Reference USER’S MANUAL ACTIVATING A DIAL SETUP ACTIVATING A LO CATION SETUP AUTO 2 DIG ITS...
Page 202 MENUS - The Audio Menu (non-mixer model) Section 11: THE WORKS – Detailed Menu Reference USER’S MANUAL AUDIO Gain T rim Send Mix Discrete AUDIO AUDIO...
Page 203 MENUS - The Audio Menu MX/MXP Section 11: THE WORKS – Detailed Menu Reference USER’S MANUAL (mixer versions) AUDIO AUDIO 1-4 Times AUDIO 1-2 Times AUDIO Assignm e nt Unlo ck e d AUDIO...
Page 204 11.2 The AUDIO Menu – Xstream See the next section, if your Zephyr Xstream has a mixer (models Xstream MX or Xstream MXP). Don't forget that you can create Location Setups to easily recall the menu selections of the AUDIO menus. Even if the ISDN line is the same, these Location Setups can be used to program these parameters (i.e. for different shows or uses in a dedicated situation. See Section 5.2 (Guided Tour of the Menus) for more on Setups. The Audio menu has items related to audio input and output –analog, AES/EBU and Livewire. HOT TIP! We’ve broken each of the menus into “pages” so you can navigate easily. For example, to jump to the second AUDIO menu you need only press the <AUDIO> button twice and you are there! AUDIO –...
Page 205 13.1) the level would be 0 dBu (nominal). The system allows for 18 of headroom above nominal level, clipping at +22 dBu. Consumer: This will be ‐11 dBu (nominal) for the factory default settings of JP10 & 11 (JP 5 & 6 on early rev audio boards). If these jumpers are reconfigured (see section 13.1) the level would be ‐15 dBu (nominal). The system allows for 18 headroom above nominal level, clipping at + 7dBu. Level Out: (Professional; Consumer) This determines the nominal output level as follows: Professional: +4 dBu (nominal). Clip point is +22 dBu. Consumer: ‐10 dBu (nominal). Clip point is +8 dBu. Gain Trim: (-12dB to +12 dB) This setting happens after the Analog‐to‐Digital converter. It will affect the levels of both the Analog and AES/EBU inputs. It can be used to adjust for differing headroom requirements when connected to other gear. Since this setting is in the digital domain changing to values greater than the default (0 dB) will decrease operating headroom, by a corresponding amount. IMPORTANT TIP! Setting the gain trim adjustment (below) to values greater than 0 dB will serve to reduce input headroom by a corresponding amount.
Page 206 • This adjustment can also be used in cases where a low‐level signal source must be used. Again you should use the rated clip point of the source device to determine the closest setting. Simply add gain to bring this rated clip point up to +22 dBu. AES sources. In the case of AES sources, this adjustment can be used to adjust system headroom. This should be done with care and deliberation. While we don’t recommend setting levels “by eye,” if you choose to do so, you can use the Send meters as your guide. AUDIO – Screen 2 INPUT SOURCE: (Analog; AES/EBU; Livewire) This selection determines which inputs are active. Your options are Analog, AES and Livewire. Note that analog, AES/EBU, and Livewire outputs are always (simultaneously) active. AES SAMPLE: This determines the sample rate of the AES/EBU Output signal as follows: Codec This option uses the sample rate selected in the Codec menu. This is the “transmission” sample rate and is derived from the ISDN (or V.35) clock. Use this option only the AES/EBU equipment being fed is tied to this network clock (for example, another codec). Sync: This option sample rate converts the codec receive audio to the rate of the AES/EBU signal present at the SYNC‐IN jack for output at the AES/EBU Out jack on the rear of the Zephyr Xstream. This can be any AES/EBU signal; any audio present will be ignored. AES Input: This is the usual setting if a house sync is not used; this option sample‐rate‐ converts the codec receive audio (set in the AUDIO menu) to the rate of the AES/EBU signal present at the AES/EBU IN jack for output at the AES/EBU Out jack on the rear of the Zephyr Xstream. This option allows easy synchronisation to your clock reference, when the Xstream's AES/EBU In is fed from a device already tied to your house clock reference. Section 11: THE WORKS –...
Page 207 “clock slip” due to different clock rates within a digital facility. Typically a “house clock” reference would be fed to the SYNC-IN jack of the Zephyr Xstream, as well as to other equipment, to prevent drop outs due to buffer “overflow” or “underflow” conditions.
Page 208 Livewire Since Livewire is one way of getting audio to and from your Xstream, we will discuss those menus here as well. Please refer to Section 8 for details on configuration of Livewire. These menus enable you to specify the Livewire streams being created by the codec (“sources” delivered to the Livewire network) and the audio streams being delivered by the codec (“destinations” from the Livewire network). Expert mode enables you to specify custom backfeeds if required. Please see Section 8 for background on Livewire and more details on the Xstream configuration. Livewire – Basic Setup The basic setup gives you the option of specifying a channel number and name for each the two stereo Livewire streams that can be created by the Xstream. The channel numbers must be unique. The name can be any name you wish to use to describe the codec. Try to choose a detailed name if you have more than one codec since this will be beneficial when you are setting up your Axia system. The Source Mode should be set to Livestream for minimum latency. The backfeeds of the codec are set automatically in this mode. They will be set for the same channel number as the source with “to source” as the mode. If you need to specify a different backfeed, you will need to use the Expert Setup shown below. Livewire – Expert Setup Section 11: THE WORKS – Detailed Menu Reference USER’S MANUAL ...
Page 209: The Audio Menu - Xstream (Non-Mixer Version)
11.3 The AUDIO Menu – Xstream MX/MXP See the previous section, if your Zephyr Xstream does not have a mixer section (model Xstream). The AUDIO menu has items related to the four main mixer inputs; local mix 1; and local mix 2. You will find many useful items in the pages of this menu. Don't forget that you can create Location Setups to easily recall the menu selections of the AUDIO menus. Even if the ISDN line is the same, these Location Setups can be used to program these parameters (i.e. for different shows or uses in a dedicated situation. See Section 5 (Guided Tour of the Menus) for more on Setups. AUDIO - Screen 1 (Status) The first AUDIO menu has the individual levels for each of the Main Mixer inputs. These levels are post fader and therefore can be used to view relative levels of each input. Clip indicators for each input are also provided. HOT TIP! The clip indicators should display very rarely, if ever. If you see the clip indicator, you should change the CHx Gain setting for the input that is clipping.
Page 210 MIC +30: ‐40 dBu (nom) Line Con: ‐10 dBu (nom) Line Pro: +4 dBu (nom) The nominal values given above assume the fader is mid‐position and the AGC/Limiter (processor) is off Ch “x” Pan This menu selection is a "panorama" adjustment that allows you to vary the amount of audio sent to the A and B codec inputs. Note that this adjustment has no effect unless the input "x" has been assigned to both A and B using the front panel control. Ch “x” Proc This menu selection allows you to enable the built in AGC/Limiter (processor) for Main Mixer input "x". You will be able to see the level for input "x" as well as the AGC and Limiter working by looking at the small bar‐graphs and gate indicator to the right side of this screen. The built in Omnia processor is quite sophisticated and offers the following options: None: This disables the input processor for channel "x". Voice L: (Voice Low) This is the least aggressive setting, intended to protect against minor level problems when running a live show. This provides AGC with no limiting. Ideal where you want the most natural sound in live situations where mixing could be difficult. Voice M: (Voice Medium) This is a moderate setting. It combines AGC with a small amount of limiting. This provides good dynamics control, and serves to "beef up" the voice. This setting is the most frequently used general purpose setting for voice. Voice H: (Voice High) This is an aggressive setting employing both AGC and limiting. This setting is perfect for easily excited talent such as sportscasters. Section 11: THE WORKS – Detailed Menu Reference USER’S MANUAL...
Page 211 HOT TIP! Please don't use the "voice" processor settings if your source includes musical material. The fast time constants used in the Voice M and Voice H settings will cause noticeable artifacts if used on most music. Music: This setting offers a slow AGC only. Low Cut This is a low cut (pop) filter designed to reduce wind and breath noise. Its characteristics are as ...
Page 212 Mon “x” Rcv Mix This parameter, in combination with Mon "x" Rcv Pan selection is useful for when only a single return channel is being sent from the far end. The default is stereo. The options are: Stereo: This choice routes the incoming (far end) audio to the headphones as received. The A channel audio appears on the Left side of the Local Mix and the B channel audio appears in the Right side of the Local Mix. A: This option routes the incoming A channel audio to both sides of the Local Mix. B: This option routes the incoming B channel audio to both sides of the Local Mix. Note that the "A" or "B" options allow a single incoming signal to be output on both channels of the Local Mix. Mono A + B: This option sums (i.e. mixes) the incoming A & B audio channels and routes this to both sides of the Local Mix. This can be particularly useful when the both a mix minus and a producer IFB (interruptible foldback) are being received, as it allows both feeds to be heard in both ears. Mon “x” Rcv Pan This control determines the ratio of the amount of the audio fed to the A (Left) and B (Right) channels of the Local Mix. Mon “x” Xmt Mix This option operates in the same was as Mon "x" Rcv mix, only the send audio is used. This is particularly useful when creating two separate mono mixes to be sent to two far end locations, as it can allow the talent to hear the correct return mix in both ears. The options are: Stereo: This choice routes the outgoing (near end) audio to the headphones just as transmitted by the codec section. The A channel audio appears on the Left side of the Local Mix and the B channel audio appears in the Right side of the Local Mix. A: ...
Page 213 B: This option routes the outgoing B channel audio to both sides of the Local Mix. Note that the "A" or "B" options allow a single outbound signal to be output on both channels of the Local Mix. Mono A + B: This option sums (i.e. mixes) the incoming A & B audio channels and routes this to both sides of the Local Mix Mon “x” Xmt Pan This control determines the ratio of the amount of the audio fed to the A (Left) and B (Right) channels of the Local Mix. AUDIO - Menu 8 This menu has three options: Phantom This option allows you to disable (Off) or enable (On) phantom power for inputs 1 & 2. Phantom power provides a full 48 volts, and is fed to both inputs 1 &2 when enabled. Level Out: (Professional; Consumer) This determines the nominal output level as follows: Professional: +4 dBu (nominal). Clip point is +22 dBu when used with balanced sources. Consumer: ‐10 dBu (nominal). Clip point is +8 dBu. Assignment This setting this to Locked locks the input assignments (A/B select) and monitor assignments (Send/Receive select) to their current settings. Section 11: THE WORKS – Detailed Menu Reference USER’S MANUAL...
Page 214: The Codec Menu
Input Select (not present in 10 Base-T version hardware) The MX/P Xstreams can be used on a Livewire Ethernet network. It is worth noting that the Livewire inputs bypass the mixer section and feed the codec directly. The input selection has the following options: Mixer: In this case, the output of the MX/P's mixer section feeds the codec. This can be considered the usual mode of operation for a MX/P Xstream Livewire: When this setting is selected, the mixer is disabled and the codecs input is fed from the Livewire channel(s) specified. Please refer to Section 8 for details on Livewire configuration. 11.4 The CODEC Menu CODEC – Screen 1: Status This page summarizes the Transmit, Receive, Bitrate & Sample Rate settings. Transmit will effect whether the far end can receive your audio Receive will effect whether you can receive audio from the far end. Bitrate and Sample Rate effect transmission in both directions Section 11: THE WORKS – Detailed Menu Reference USER’S MANUAL...
Page 215 CODEC – Screen 2 Transmit: The Transmit setting determines what standard (i.e. algorithm) is used to encode the audio to be sent to the far end. Each mode offers distinct advantages and disadvantages (see section 6.2 for additional information), as follows: L3 Stereo and L3 J‐Stereo Until the release of the Zephyr Xstream, MPEG Layer‐3 was the best available algorithm for sending stereo audio over 2 ISDN B channels. Joint (i.e. Joint Stereo) should be used for the majority of applications, as it eliminates the redundant transmission of the center channel information typical in most source material. L3 Stereo is appropriate for situations where the audio transmitted will be subject to playback over Surround Sound systems (i.e. mostly for Television). Use these Layer‐3 stereo modes if the far end does not support AAC. This mode offers 15 kHz audio frequency response if the Sample Rate is set to 32 or 16 kHz audio frequency response if the Sample Rate is set to 48 kHz. Section 11: THE WORKS – Detailed Menu Reference USER’S MANUAL...
Page 216 L3 Mono 128 This is the best quality Layer‐3 mode supported on the Xstream. Audio present on the A input will be encoded and transmitted. By using both B channels (i.e. 128 or 112 kbps) reduces the compression ratio by half and therefore gives superior fidelity. Since two B channels are required however, the transmission costs would be doubled. If lengthy sessions are to be performed over long distances, a mode that requires only a single B channel may be desirable. Use L3 Mono 128 when mono transmission is needed, transmission costs are secondary, and the far end does not support AAC. This mode offers 15 kHz audio frequency response if the Sample Rate is set to 32 or 22 kHz audio frequency response if the Sample Rate is set to 48 kHz. ZEPHYR CLASSIC VS XSTREAM COMPARISON! While the Zephyr Classic cannot transmit Layer 3 Mono-128, it can receive audio coded in this way. Use the L3 Stereo Rcv mode to do so. L3 Dual/Mono ...
Page 217 site X will get the audio from audio input A and site Y will get the audio input into audio input B. • Finally, the mode can be used to transmit two channels between two units. Both "lines" must be dialed up – Line 1 to Line 1, Line 2 to Line 2. It is important to understand the difference between this use of L3 Dual/Mono and the L3 Stereo and L3 J‐Stereo modes. a) L3 Dual/Mono keeps the two audio channels completely separate. In fact, it does not ensure that the two channels remain phase‐ aligned. b) L3 Stereo keeps the two channels separate (i.e. there is no danger of cross‐talk) and maintains phase‐alignment between the two channels. However, the perceptual model used with L3 Stereo does assume that the listener will be hearing both channels simultaneously (i.e. that this is a stereo application). Therefore, the model takes into account that audio from one channel may mask audio from the other channel. c) L3 J‐stereo matrixes the two channels together and therefore a small amount of inter‐channel interaction is possible. L3 Dual offers 15 kHz audio frequency response when the Sample Rate is set to 32 kHz, or 16 kHz audio frequency response if the Sample Rate is set to 48 kHz. ZEPHYR CLASSIC VS XSTREAM COMPARISON! Early Layer-3 Zephyr’s call this mode "Mono/Dual". Later Layer2+3 Zephyrs call this mode "L3 Dual".
Page 218 In Layer-3, the Zephyr Xstream uses the FHG/Telos Zephyr IMUX protocol. Other Layer-3 codecs support this method, allowing for compatibility among different codec manufacturers. The Telos IMUX used in AAC and AAC-LD is available on our website to encourage compatibility with other manufacturers that may choose to support AAC in the future.
Page 219 COMPTIBILITY TIP! The Zephyr Xstream cannot receive two stereo feeds from the far end (after all, where would the audio be output?). When connected to two different sites simultaneously, you can receive an audio feed back from each using G.722 or L3 Dual/Mono.
Page 220 G.722 acts similarly to the Dual modes discussed above. Audio input A is encoded and transmitted on "line 1" while Audio input B is transmitted over "line 2". Offers 7.5 kHz audio frequency response. Delay is very low at 14 milliseconds. L2 Stereo L2 J‐Stereo These work similarly to the equivalent Layer‐3 modes. Two connections are required. L2 Stereo offers discrete stereo transmission with 7.8 kHz audio frequency response when the 48 kHz Sample rate is used if the transmission channel is 56 kbps. It offers 9.8 kHz frequency response when the 48 kHz Sample rate is used if the transmission channel is 64 kbps. L2 J‐Stereo offers 20 kHz audio frequency response when the sample rate is 48 kHz and 15 kHz response when the sample rate is set to 32 kHz. HOT TIP! Of course you will want to use Layer 3 or AAC when supported at the far end. We strongly recommend Layer 2 Joint stereo over Layer 2 Stereo when transmitting stereo material in Layer 2 at 128kbps.
Page 221 In Layer-2, the Zephyr Xstream uses the CCS CDQ (TM) IMUX protocol. Most Layer-2 codecs support this method, allowing for compatibility among different codec manufacturers. The Telos IMUX used in AAC and AAC-LD is published on our web site to support interoperability with other manufacturers implementing AAC and AAC-LD L2 Mono 64 ...
Page 222 It will be helpful to understand the following terminology when determining what Receive mode should be used: 64 ‐ This is a Zephyr Xstream Transmit mode that requires only a single B channel to operate. The actual bit rate of that channel could be either 56 or 64 kbps. The correct Receive mode will also have "64" in it (for example "L2 64"). 128 ‐ This is a mono Transmit mode that requires two B channels (i.e. calls) to operate. The actual bit rate of those channels could be either 56 kbps (for a total of 112 kbps) or 64 kbps (for a total of 128 kbps). The correct Receive mode for a Transmit mode with "128" in the name will just have the name of the algorithm (for example "L2"). Dual/Mono ‐ This is simply a special case of Mono 64. In the Dual Mono case the A audio Input source is sent over "Line 1" while the B audio Input source is sent over "Line 2". J‐Stereo ‐ This is "Joint Stereo" and is the preferred stereo mode in most cases. The biggest exception is when stereo material will later be transmitted over a surround sound transmission system (i.e. for Video or Film production) in which case the Stereo mode should be used. All stereo modes require 2 B channels (2 calls) unless otherwise stated. The correct Receive mode for a Transmit mode with "J‐Stereo" in the name will just have the name of the algorithm (for example "L2"). Stereo ‐ This is "Discrete Stereo". The major use for this is when stereo material will later be transmitted over a surround sound transmission system (i.e. for Video or Film production) in. All stereo modes require 2 B channels (2 calls) unless otherwise stated. The correct Receive mode for a Transmit mode with "Stereo" in the name will just have the name of the algorithm (for example "L2") unless otherwise stated. J‐Stereo 64 ‐ This mode, unique to AAC, is the only stereo transmission mode that does not require 2 B channels. Half 64 ‐ This is a special Layer‐2 mode that operates at 24 kHz sample rate. The audio will be transmitted at 24 kHz sample rate regardless of the actual sample rate selected in the CODEC menu. The Xstream's Receive modes are: L3 Receives all Layer‐3 modes except for the L3 Dual/Mono mode (and L3 mono streams at 56 or 64 kbps from other codecs). L3 Dual/Mono Can receive 1 or 2 L3 Mono streams at 56 or 64 kbps from other codecs. The audio bit stream connected to "Line 1" will be output on audio channel A, while the stream received on "Line 2" will be output on audio channel B. L2 Receives all Layer‐2 modes except for L2 mono 64. ...
Page 223 L2 Mono 64 Receives L2 Mono 64 (at 56 or 64 kbps) over "line" 1. G.722 Receives G.722. Zephyr Xstream uses the Statistical Recovery Timing (SRT) synchronization method. This is a "Dual" mode and behaves the same way as the L3 Dual/Mono mode. The audio bit stream connected to "Line 1" will be output on audio channel A, while the stream received on "Line 2" will be output on audio channel B. AUTO This mode will attempt to determine the mode being sent from the far end and internally set the Receive mode to match. The Sample Rate must be matched for this to occur. As the auto mode takes several seconds to determine the correct mode, there are times you may wish to use a specific setting for Receive. In this case, your receive mode must be set to decode the stream being sent from the far end. You will need to know the Transmit (“Xmt” or “Encoder”) mode of the far end to set this correctly. HOT TIP! The system will display the mode that was found by the AUTO algorithm on the default screen as well as the CODEC Status screen. You may wish to change your Receive setting to that mode.
Page 224 For more about this option when used with those Interface selections see Appendix 6 (Special Operational Modes) COMPATIBILITY TIP! The Zephyr Xstream specifies the bit rate per channel. Since certain modes (i.e. stereo or mono-128) require two channels to operate, the total (aggregate) bit rate would be double the number displayed in these cases.
Page 225 IMPORTANT TIP! 1) Zephyr Xstream only supports the 24kHz sample rate in MPEG Layer-2 (or when used in “IP streaming mode” where the selections for this menu are different). Therefore, the Sample Rate menu does not have this option. The “L2 Half 64” Transmit mode is used to achieve 24kHz operation in MPEG Layer-2.
Page 226 COMPATIBILITY TIP! Not all codecs support all sample rates. 32kHz is most commonly supported in MPEG Layer- 3 whereas 48kHz is more commonly used in Layer-2. 24kHz is sometimes seen in Layer-2. In this case set your Transmit or Receive to “L2 Half/24”...
Page 227: The Tel Menu
11.5 The TEL Menu TEL – Screen 1: Status In ISDN or V.35 mode… This screen shows the status of each of the two ISDN B channels. The status will vary depending on the Interface selected in the ISDN Menu, and can be as follows: Status words when ISDN Interface is selected: Inactive – This indicates that neither the built‐in NT1 nor the S interface is active. This will be displayed if no ISDN line is present. It is also displayed on ETS 300 (Euro‐ISDN) lines after periods of inactivity (the ETS 300 standard allows the central office switch to put the line to sleep). SPID Pending – This option is only valid when the Telco selection of the TEL menu is set to “Natl.I‐1” or “DMS Cust.”. This indicates that the Xstreams internal NT1 has achieved "U synchronization" with the BRI circuit, or that the S interface connection to the external NT1 is active. This also could be displayed if the BRI is grossly misconfigured. Section 11: THE WORKS – Detailed Menu Reference USER’S MANUAL … ...
Page 228 SPID Error ‐ The SPID entered for the indicated channel is incorrect. You may have entered it wrong, or the Telco may have given you the wrong SPID (or entered the SPID incorrectly in their switch). Ready – This indicates that an ISDN line is present. It indicates that all handshaking with the central office is complete and the line is ready to be used. Note: If Ready is displayed and the Xstream cannot make or receive calls you may have the Telco option set incorrectly or you may have omitted your SPIDs. Conn: xxx ‐ The channel is connected to "xxx". "xxx" is the caller ID of the calling party if available for calls originated from the far end, or the number dialed for calls originated by the Zephyr Xstream. A call timer below, the line status, displays the length of time the call has been connected in the following format: Days‐hours:minutes:seconds Pending Seize ‐ Outgoing call in progress; Request for call has been sent to the Telco switch and Zephyr Xstream is awaiting acknowledgment. Seized ‐ Outgoing call in progress; Telco switch has acknowledged the call request. Proceeding ‐ Outgoing call in progress. Telco switch has acknowledged the call request and is waiting for routing information from the network. Outgoing Ring ‐ Outgoing call in progress; far end has generated a call alert signal. Ring: xxx ‐ Incoming call from "xxx" has been detected and is being processed. "xxx" is the caller ID of the calling party, if available. Accepting Call ‐ Incoming call is being answered Disconnecting ‐ Request to disconnect has been made. Awaiting acknowledgement. Ended ‐ Request to disconnect has been made and the call has been cleared. Inactive: The internal NT1 has not initialized. External NT1 (if any) is not active on the S interface. Also displayed on ETS300 lines that have been "deactivated" or put to sleep. Note this case is not unusual, and lines in this case will reawaken when used. Note, this may also be displayed if the Telco setting in the TEL menu is set incorrectly. ISDN TIP! If you reboot the Xstream and the line status shows Ready & Ready, but you cannot make or receive calls, check the following: -Is the Telco selection in the TEL menu set correctly? Certain incorrect combinations may cause a false Ready indication.
Page 229 ISDN TIP! If the line status shows Ready & Ready and you can make calls, but cannot receive calls, check the following: -Is the Telco selection in the third TEL menu set correctly? If you have DMS Cust. ISDN (USA & Canada only) and your Telco is set to Natl. I-1 you may be able to place calls outgoing calls only.
Page 230 USER’S MANUAL Status Screen when Ethernet Interface is selected: The first Tel screen shows the number of attached decoders (players) whenever the Interface is set to Ethernet. TEL – Screen 2: SPID 1 This is your first SPID (Service Profile Identifier) number. SPIDs are only used in the USA & Canada. If you have ETS 300, INS 64 (Japan) or AT&T Cust. (PTP) for your ISDN protocol, you should leave this blank. When you are using Natl. I‐1 or DMS‐100 for your Telco mode (see below), you must enter the correct Service Profile ID (SPID) numbers for the ISDN BRI here and below. Normally, the Telephone Company tells you this number when they confirm your order. Often it is found written on the line jack or punch block. SPID1 corresponds to Line 1. Usually, the SPID is the area code+phone number+0101. However, we have seen some that include a two‐digit prefix and a two‐digit suffix, and many other variations are possible. See Appendix 4 for a list of known working SPIDs by Telephone Company. Despite the last paragraph, you cannot assume anything about the SPID! If the above advice works, consider yourself lucky; if it doesn’t work, there is no substitute for getting the correct SPID from your Telco! Don’t let the installer depart without leaving them. You’ve been warned! For more on SPIDs see Section 7 (ISDN Basics) and Appendix 7,8 (Ordering ISDN). SPID 2 This is the SPID for your second ISDN B channel. See above. Section 11: THE WORKS – Detailed Menu Reference...
Page 231 In an emergency, to save a remote, you might be able to get a line to work without proper SPIDs by dialing into it. If it works, this fix is only temporary, and the line will fail to initialize next time the Zephyr Xstream is booted. ISDN TIP! If you see SPID ERROR as your line status, your SPID for that channel is incorrect.
Page 232 ISDN TIP FOR BELLSOUTH CUSTOMERS! Despite the fact the BellSouth originated the proposal for the standardised SPID (see above), many regional Bell South offices leave off the last two digits of the SPIDs when giving them to customers. If you are a BellSouth customer and are given a SPID of the format: Area Code + 7 digits + 01 and you get "SPID Error"...
Page 233 TEL – Screen 3: Telco: This is your ISDN protocol. This must be set correctly for the Xstream to make or receive calls. Select the correct value for your ISDN BRI circuit. Section 7 (ISDN Basics) has additional background information on ISDN. The correct setting will be determined by the type of Central Office switch providing your ISDN service. Note that some switches offer more than one protocol. See below for a table showing the correct settings for commonly used switches. TELCO SWITCH BRI LINE PROTOCOL AT&T/Lucent 5ESS AT&T Custom Point to Point AT&T/Lucent 5ESS National ISDN-1 National ISDN-2 NORTEL DMS-100 Custom Functional NORTEL DMS-100 National ISDN-1 National ISDN-2 JAPAN NTT Ins 64 EUROPE &...
Page 234 ZEPHYR CLASSIC VS XSTREAM COMPARISON! Zephyr (classic) and ZephyrExpress use the Telco option "Natl I-1" for both the National ISDN protocol and the DMS Custom functional ISD protocol. Zephyr Xstream distinguishes between these two protocols so the correct choice must be made. Auto Ans.
Page 235 All calls, regardless of call types, will be answered automatically. ISDN Circuit Switch Data calls (i.e. callers from a Zephyr Xstream when the Call Type is set to “Zephyr”) will be answered automatically. Circuit Switched Voice (i.e. POTS/G.711) calls are treated as if “none” was selected. Phone Circuit Switch Voice calls (i.e. callers from an Xport on a POTS line, a regular telephone, or a Zephyr Xstream when the Call Type is set to “Phone”) will be answered automatically. Circuit Switched Data calls are treated as if “none” was selected. Interface The Interface selection will activate one of the following ports of the Zephyr Xstream ISDN/ISDN Modem Either of these selections will enable the Zephyr Xstream's built‐in ISDN interface, if present. The ISDN Modem selection loads the code required to talk to the Zephyr Xport connected to a POTS (analog) line. When call is placed in Phone Mode (in the DIAL menu), the Interface setting will change between ISDN to ISDN Modem automatically. This will also occur when an incoming "voice" call is answered, and can be ignored by the user; if an ISDN mode call is subsequently made, or received, the unit will switch back to ISDN interface automatically. A note regarding Mode detection and Interface switching of incoming calls: The ISDN network will signal if a call is "voice" or "data" call. Since modems use POTS technology, calls from Xport are considered "voice" calls. In the case of a "data" call, no other information is needed; the Xstream will switch from the ISDN Modem Interface to ISDN, if necessary, and answer the call. If the network indicates an incoming call is a "voice" call, the Xstream must determine if this is a call from a telephone, or from the modem of an Xport. Therefore, the Xstream switches the Interface selection from the ISDN to ISDN Modem, if necessary, answers the call, and then listens for a modem carrier tone. If no modem tone is detected, the unit un‐mutes the audio and the call is considered a "Phone" Mode call. If a modem tone is detected, the Xstream's modem will begin negotiating with the Xport. Therefore, all incoming "voice" calls have a short delay before audio is heard. V.35 Split This selection will enable the Xstream's built‐in V.35/X.21 interface, if present. Section 11: THE WORKS – Detailed Menu Reference USER’S MANUAL...
Page 236 The V.35 Split works the same way as the V.35 mode on the Zephyr classic. The same Inverse Multiplexing (IMUX) method will be used as if the Interface were set to ISDN. The codec selections in V.35 Split are also the same as for ISDN. Note that the line status words on the default Tel menus have different meanings when V.35 is selected (see above). See Appendix 6 (Special Operational Modes) for more information. V.35 Single This selection will enable the Xstream's built‐in V.35/X.21 interface, if present. Section 9 (V.35/X.21) has more details on the operation of the V.35 interface. In Single V.35 mode a single MPEG compliant bitstream is generated and output over both V.35 ports. Note that the Line status words on the default menu and the Tel menu have different meanings when V.35 is selected. See Appendix 6 (Special Operational Modes) for more information. IMPORTANT! The following selections in the CODEC menu will have different options when the Interface is set to V.35 Single. • Transmit • Receive • Bit Rate See Appendix 7 (Special Operational Modes) for additional Information.
Page 237 Ethernet (SIP) and Ethernet (RTP) This selection enables Zephyr Xstream's built‐in Ethernet interface for audio coding and/or decoding. This allows for transmission and reception of audio over Internet Protocol (IP) based networks. The (SIP) and (RTP) designations refer to what protocol is used to establish communication. SIP (Session Initiation Protocol) connections are bi‐directional. By ‘dialing’ another Zephyr Xstream, you request a return feed as well. RTP (Real‐time Transport Protocol) is a mono‐directional, ‘push only’ mode. When you ‘dial’ another Zephyr Xstream in RTP mode, you send audio to the remote site, but receive nothing in return unless the remote operator manually initiates a connection back to the studio. Appendix 6 covers this topic in detail. IMPORTANT! The following menus will have different options when the Interface is set to Ethernet. • CODEC • DIAL Prefix: This number is appended to the beginning of every number dialed. Useful when a "9" or other digit is required for access. DEEP TECH NOTE! Long Distance services that rely on "in-band"...
Page 238 TEL - Screen 4 This screen has setup information that is only needed when using on off the Ethernet options for Telco (see above). For more information see Appendix 6, Special Operational Modes. IP ports to be used by the streaming system are listed on the Telco configuration page in all modes. (HTTP page shown below) SIP Port (Default 5060) SIP uses a TCP port, as its communication is bidirectional. The SIP port is what is used for the connection protocol, that layer that initiates the two RTP streams. The SIP port must match on both ends of a connection. UDP Port (Default 9150) RTP streams are sent and received over the UDP port. This port is used for both RTP and SIP modes, as SIP negotiates the creation of RTP streams. The UDP port must match on both ends of a connection. HTTP Port (Default 8080) The TCP port used for HTTP streaming only (normal web‐pages are also served over HTTP, but these are sent over port 80, the usual port to which a web browser connects). A player which supports HTTP streams can connect using the usual port notation (e.g. http://www.xxx.yyy.zzz:PORT). Note that prior to 3.0.1p, the Zephyr Xstream used port 8000 for HTTP streaming. For stability, this streaming option should only be used for debugging, not on‐air delivery or monitoring. TCP Port (Default 8800) This port is currently unused. Section 11: THE WORKS – Detailed Menu Reference USER’S MANUAL ...
Page 239 TEL – Screen 5: DTMF: (On; Off) This enables/disables DTMF "Touch Tones" when connected on a voice call (i.e. call type = "phone"). These are useful for navigating through voice menu systems. We offer the option of turning this off in the case that you may be entering information with the keyboard while dialed up on a voice call. Redial Sets the delay between automatic ISDN redials, or disables the feature. If an outgoing call is active when the Xstream loses ISDN connectivity or power, it will attempt to re‐ establish the call as soon as the error state is cleared (that is, when the ISDN comes back to life, or power is restored to the Xstream). You can adjust the time between these attempts, or disable them entirely. The redialing feature is also used when attempting to dial via contact closures and v.35 roll‐over (see next item). ISDN Backup On an Xstream with both a v.35/x.21 card and an ISDN card, you can enable a rollover in case of v.35/x.21 failure. If you choose to enable this feature, this menu item will allow you to select an dial setup (any of your first ten) to be used. Note that by enabling this feature, you also allow ISDN calls to switch the Xstream from v.35/x.21 mode to ISDN mode (This is necessary because while one side of the high speed serial connection may see a failure, the other side could be operating as usual. In this case, when the side seeing the failure condition switches to ISDN mode and redials, the other side must respond to the ISDN request). Section 11: THE WORKS – Detailed Menu Reference USER’S MANUAL...
Page 240: The System Menu
11.6 The SYSTEM Menu SYSTEM - Screen 1 This menu gives summary and troubleshooting information about the system: • Uptime ‐ The times since the system was last powered up or rebooted. • Loopmode ‐ The currently selected loop mode set in the SYSTEM menu (see below). • Version ‐ Software version running on the unit • Interface ‐ The currently selected Interface in the TEL menu. SYSTEM - Screen 2 This menu has the most frequently used System menu items. These are: Section 11: THE WORKS – Detailed Menu Reference USER’S MANUAL...
Page 241 Contrast This allows you to change the contrast of Xstream's LCD display to accommodate differing room lighting or viewing angles. Volume No, this doesn't belong in the AUDIO menu! This volume selection is to adjust the volume of the different sounds the Xstream makes when keys are pressed, calls connect or other things happen. rs232 This allows you to easily set the bit rate of the Zephyr Xstream's RS‐232 port on the rear panel. Rates from 300 bps to 57.6 kbps are supported. The other serial port parameters are: • Data Bits: 8 bits • Parity: None • Stop Bits: 1 • Flow control: none About This gives you information about your Xstream. SYSTEM - Screen 3 Pressing the SYSTEM button a third time shows this screen with the selections needed to configure your IP (Internet Protocol) settings. You won't need to enter this information to operate the unit from the front panel. You will need to enter this information if you will be connecting to the Xstream's Ethernet port (i.e. will be using the Ethernet mode, or will be controlling the unit remotely using this port). You will also need to enter these items when the time comes to update the Xstream's operating software. Section 11: THE WORKS – Detailed Menu Reference USER’S MANUAL ...
Page 242 USER’S MANUAL IP Addr This is the IP address for this Xstream. As with any computer on an IP network, the Zephyr Xstream must have an IP address before it can be used over the network. This address must be unique, so you’ll have to ask whoever manages your network to give you this number. This is a "fixed" or "static" IP address. If you plan to use the web browser to setup your Xstream, you will first have to set its IP address here. You may connect the Xstream to your LAN or use a crossover cable to your PC. Make sure the PC also has an IP address in the same range as the Xstream. Actually, an IP address is a series of 4 numbers separated by periods. This is a numeric‐only field; enter the numbers using the dial pad. Both # and * will give you a period. As usual, the 5 key will backspace if you make a mistake, or need to delete an old entry. SN Mask This is your subnet mask. The default 255.255.255.0 is usually fine. Change this only if your Network administrator tells you to. This is also an IP address, see the rules above. The subnet mask is to determine the size of your “local” network. All packets addressed to a destination outside this “local” area are sent to the gateway node entered in the next selection. Just as with the IP address, the Subnet mask is a series of 4 numbers separated by periods. Enter the numbers using the dial pad. Both # and * will give you a period. This is a numeric‐only field; enter the digits as described above. As usual, the 5 key will backspace if you make a mistake or need to delete an old entry. Section 11: THE WORKS – Detailed Menu Reference...
Page 243 Gateway This is needed only when using IP to communicate with other equipment outside your local network. It is required if you are downloading new software directly from the Telos FTP site. This is also an IP address, see the rules above. This is the IP address of a gateway router connecting you to the internet. It is an IP address just as with the previous items and is entered in the same way. You will need to get this from your network administrator or internet service provider. This is a numeric‐only field; enter the digits as described above. DNS is needed only when downloading software directly from the Telos FTP site. This is an IP address, see the rules above. This is the IP address of the DNS (Domain Name Server) you will be using. It is an IP address just as with the previous items and is entered in the same way. You will need to get this from your network administrator or internet service provider. HOT TIP! If you need to "borrow" the IP address of a Windows computer to use when updating your extreme, here are some tips: 1. Go to the MS-DOS command line of the Windows machine (Run " cmd <enter>" for Win 2000 or Win NT;...
Page 244 Not that entering anything in this *FTP Site This setting is used for updating earlier units. If this option is not present you have the later unit with the 100 Base‐T Ethernet and Livewire capability. See Section 13 for how to upgrade the Xstream. This is the site to which the Xstream will download new software from. If your firewall allows FTP access, you can use the default which is ftp.telos‐systems.com . Or, you can enter the IP address of a local FTP server. This can be an IP address (see the rules above) or it can be an alphanumeric entry. DEEP TECH NOTE! When an alphanumeric entry is present, a domain name server (DNS) is queried for the IP address of the computer using that domain. The reason we do so is that while the Zephyr.com domain will remain constant, the IP address of our FTP server (ftp.zephyr.com) might very...
Page 245 3. You should receive a reply. If you do not get a reply check with your IT manager for assistance - your firewall may not allow FTP access. The reply will include the current IP address of the Telos FTP server. 4. Enter this IP address as "FTP site" in the Xstream.
Page 246 HOT TIP! The AUTO Receive mode (in the CODEC menu) may not work in the ISDN loop mode depending on the INTERFACE option selected (in the TEL menu). Therefore, we suggest manually setting the Receive mode when performing tests with this option.
Page 247 USER’S MANUAL Restore Factory Defaults... This selection allows you to clear all settings of your Xstream. All menu options will be returned to the factory default settings. Don't use this option unless you are sure you wish to erase these settings. You will be prompted with a WARNING screen inquiring if you really wish to erase all your settings. Press the <SEL> button (or any of the MENU buttons) if you wish to abort. If you choose to proceed with erasing your settings, select OK by pressing <6> or <5>to select OK. Then press <SEL> to proceed. In rare cases where the Zephyr Xstream's internal memory has become corrupted, the Restore factory defaults selection may help clear up the misbehavior of the unit. SYSTEM - Screens 6 & 7 SYSTEM Menus 6 & 7 determine the function for each of the eight parallel inputs of the serial port. There are eight inputs, numbered 0 through 7. Each input can be set to any of the following values. Section 11: THE WORKS – Detailed Menu Reference...
Page 248 PP in 0 to PP in 7 End‐to‐End This option is used to pass contact closure information to the far end using the ancillary data channel present in certain codec modes (see Sections 12.1, 3.2 & 3.4). HOT TIP! An ancillary data channel will only be present in a given transmission direction if the receive codec is "locked" and one of the following codec modes is employed in that direction. AAC (all modes), AAC-LD (all modes), Layer 3 (all modes) Layer 2 mono 64 ZEPHYR CLASSIC VS XSTREAM COMPARISON! NOTE:...
Page 249 SYSTEM - Menus 8 & 9 PP Out 0 to PP Out 7 SYSTEM Menus 8 & 9 determine the function for each of the eight outputs of the serial port. The option for reach output can be separately set to any of the following values. For more information on the Parallel port hardware interface see Sections 3.2 (Guided Tour of the Hardware ‐ Non Mixer Version) and Section 3.3 (Guide Tour of the Hardware MX/P). For more information on Livewire GPIO and how it applies to the Xstream’s parallel ports, please see Section 8: LIVEWIRE – IP Audio. ON Forces this output to the "true" state. This will cause the input to sink current thereby activating an external device. This mode is intended for troubleshooting. OFF Forces this output to the "false" state. Input will become high impedance with regards to ground and will "float" to 5 volts due to the built in pull up resistor. This mode is intended for troubleshooting. Mixer On (MX and MXP only) Causes this output to go the "true" state whenever one of the front‐panel mixer inputs is assigned to A, B or Both channels. This can be connected to a local "on‐air" signal or can be connected to an input to pass this indication to the far end. 1 AND 2 Conn Causes this output to go the "true" state whenever both Line 1 and Line 2 are in the "Connected" state (as indicated on the first TEL menu). 1 OR 2 Conn Causes this output to go the "true" state whenever either Line 1 or Line 2 or both are in the "Connected" state (as indicated on the first TEL menu). Line 2 Conn Causes this output to go the "true" state whenever Line 1 is in the "Connected" state (as indicated on the first TEL menu). ...
Page 250 Line 1 Conn Causes this output to go the "true" state whenever Line 2 is in the "Connected" state (as indicated on the first TEL menu). Lock A* AND B* Causes this output to go the "true" state whenever both Lock A and Lock B are "on" (as indicated on the corresponding front panel LEDs). Lock A* OR B* Causes this output to go the "true" state whenever either Lock A or Lock B or both are in the "on" state (as indicated on the corresponding front panel LEDs). Lock B* Causes this output to go the "true" state whenever Lock A is in the "on" state (as indicated on the corresponding front panel LED). Lock A* Causes this output to go the "true" state whenever Lock A is in the "on" state (as indicated on the corresponding front panel LED). * Some Xstreams have a "Lock 1" and "Lock 2" LEDs while others have "Lock A" and "Lock B" LEDs. End‐to‐End Causes this output to follow the state of the corresponding Input of a Zephyr to which it is locked. When no connection is present (or a coding mode is used that does not support ancillary data), this output will go to the false state. Section 11: THE WORKS – Detailed Menu Reference USER’S MANUAL...
Page 251: Special Menus: Auto, Dial, & Drop
11.7 Special Menus: Auto, Dial, & Drop Step by step instructions on using these keys are located in Section 4.2. In this section we will go into additional details about the DIAL menu and will present advanced information about how the Dial Setups work under special circumstances. DIAL menu The DIAL menu appears after the Dial button has been pressed and looks like the following: If you wish to dial a data call to another ISDN codec (Mode = Zephyr) you can simply enter the desired number and press <DIAL> again. A second call can be placed on "Line 2" by pressing <DIAL> a third time. Pressing <DIAL> twice, without entering a number, will connect to the last number dialed on that line, in the same Mode used for the previous call, making it easy to reconnect if accidentally disconnected. To drop all calls in progress, simply press <DROP> four times. This sequence is illustrated here. Section 11: THE WORKS – Detailed Menu Reference USER’S MANUAL Ready Ready DIAL Phone # DIAL Connected: 0:00 Connected: 0:00 DROP Four Times...
Page 252 IMPORTANT TIP! The Number field in the DIAL menu is different from other entry fields in the Zephyr Xstream. It is NOT necessary to press <SEL> before entering the number to be dialed. While we like consistency, this field is used so often, it made sense to make it easier by doing this.
Page 253 DROP menu The Drop menu is activated by pressing the <DROP> button once. It looks like this: The system will default to Line 1 if that line is in use. It will default to Line 2 only if Line 1 is not in use. Press <DROP> a second time to drop the line shown. To drop a Line other than the line shown, press the <SEL> button, then press <5> or <6> to select the desired Line(s). To drop the Line(s) press <DROP> again. To easily clear both ISDN lines simply press <DROP> four times. AUTO menu The AUTO menu is used to enter or activate Dial Setups and Location Setups. Dial Setups Dial Setups are very similar to the "Auto Dial Setups" used in the classic Zephyr. Each of the 100 dial setups consists of several pieces of information. See Section 4 (Basic Operation and Installation) for information on creating and using Setups. • A Setup number – this 2 digit number (00 to 99) can be used to instantly access a given setup. • An alphanumeric Name • Number 1 (optional) – A telephone number to be dialed on “Line 1” • A “mode” for number 1 – Either Zephyr for a codec call, Xport for a call to a Zephyr Xport using a POTS line, or Phone for calls to a standard telephone. • Number 2 (optional) – A telephone number to be dialed on “Line 2” • A “mode” for number 2 • Include selection. This allows you to include CODEC menu settings (optional) – This is a snapshot of all settings of the CODEC menu. This allows the unit to automatically change to the settings used with a given site based on the Dial Setup used. Section 11: THE WORKS – Detailed Menu Reference USER’S MANUAL Connected Connected...
Page 254 V.35 Split, or V.35 single, the system will change the interface to ISDN so that the number(s) may be dialed. ZEPHYR CLASSIC VS XSTREAM COMPARISON The Zephyr Xstream’s Dial Setups offer some important differences from the Zephyr’s Auto Dial Setups: 4. Inclusion of codec information is optional with the Xstream, whereas it is mandatory with the Zephyr.
Page 255 Dial Setups when using V.35/X.21 Dial setups are sometimes used as a "short‐cut" method of changing codec parameters when using the V.35 option. If you are using one of the Xstream's V.35 interface modes (see Section 9 ‐ V.35/X.21 Interface) you can use Dial Setups without phone numbers to conveniently changes modes. When a setup containing an entry for Number 1 or Number 2 (i.e. a phone number) is activated and the Interface (in the third TEL menu) is set to V.35 Split or V.35 single the system will change the interface to ISDN so that the number(s) may be dialed. This arrangement allows for externally triggered ISDN backup in case of a failure of to a dedicated data facility. The Panic Dial feature could be used to activate this setup by an external device. HOT TIP! You are combining two features in this case, so be sure to understand the following: -There is no method to use a setup to change from ISDN to V.35. -Therefore, your external panic dial trigger source should remain asserted.
Page 256 USER’S MANUAL HOT TIPS! Users have found the following “Panic Dial Tricks” handy: 1. Mode reset. Create the Dial Setup(s) to be used without any phone numbers and to "include" codec information. Name them after the associated modes stored with the setup(s). You can now use push a momentary switch connected to the panic dial input(s) and Xstream will switch to a designated mode ready for an incoming call.
Page 257: System Functions And Remote Control
12 System Functions and Remote Control 12.1 Ancillary Data Serial Data Transmission The serial interface can be used to control the Zephyr Xstream, i.e. to change transmission modes or request status information. A Telnet session can also be used to perform this function. The serial interface can be switched to transparent or ancillary data mode using the atdt command. While the transparent mode is active, all data sent to the Zephyr Xstream's serial input is inserted into the coded audio signal and – provided there is an valid ISDN, V.35, or IP connection – transmitted to the receiving Zephyr. The latter will extract ancillary data information from the coded audio signal and output it on its own serial interface if it is in transparent mode. • The ancillary data transmission is fully bi‐directional and does not affect any ongoing audio transmission. The necessary bitrate to transmit ancillary data, however, is taken from the overall available line bitrate. • Ancillary data that is fed into the Zephyr Xstream without having a connection will be lost. • In Dual/Mono transmit mode ancillary data is transmitted to both sites. In Dual/Mono receive mode ancillary data line 2 is ignored. • The ancillary data channel is an on‐demand service, which means that the audio bitrate is only reduced if necessary. However, data can be transmitted continuously at up to 9600 kbps (Stereo and Mono‐128) or 2400 kbps (Mono‐64 and stereo‐64). See below. • Ancillary data is only available in certain modes, as outlined below: Section 12: System Functions and Remote Control USER’S MANUAL...
Page 258 TRANSMIT MODE L3 Dual/Mono L3 Stereo/Jstereo L3 Mono 128 AAC Mono 64 AAC-LD Mono 64 AAC Stereo/Jstereo AAC-LD Stereo/Jstereo AAC Mono 128 AAC-LD Mono 128 L2 Mono 64 L2 Half 64 Ancillary data is supported in these modes when the Interface option in the third TEL menu is set to ISDN, Ethernet, or V.35 split. When set to V.35 Single, Ancillary data operates at 9600bps. IMPORTANT! When the Interface option in the third TEL menu is set to V.35 Single ancillary data is possible in all permissible transmit modes at 9600 bps.
Page 259: Remote Control
Contact Closures For additional information on connecting to the parallel GPIO port see manual Sections 3.2 (Guided Tour of the Hardware ‐ non mixer) and Section 3.4 Guided Tour of the Hardware MX/P and Section 11.6 ‐ 11.7 (Detailed Menu Reference). Parallel outputs may be affected either by the corresponding input at the remote Zephyr or Xstream (end‐to‐end operation), or can be used to monitor functions of the Xstream. See Section 11.6 for details on the menu selections affecting the contact closures functionality. Normally, the closures are simply passed‐through from one end to the other. The function of each input or output can be set using the front panel: 12.2 Remote Control The Command Line Interface The command line interface can be accessed in two ways: Via the RS‐232 serial port or by using Telnet to access the unit over IP through the serial port. RS‐232 Port The RS‐232 serial port can be used for computer control of the system. Full status and control are available via this port, including ISDN dialing, storage of set‐ups, etc. The communication protocol is simple ASCII text, so it may be used easily with any computer and terminal emulation software. Some are using a palm top computer with macros to control their Xstreams in the field Some “terminal emulation” software possibilities are: • For the PC: Crosstalk, Procomm, MS Windows built‐in HyperTerminal, etc. • For the Macintosh: Z‐Term, White Knight, etc. The Zephyr Xstream may be connected locally to a personal computer, or to a modem for control from a remote site. There will be differences in the required cabling, as described in the Section 3. The Zephyr Xstream serial port is configured as DCE (Data Terminal Equipment), which means it looks like the connection on a modem, rather than on a computer. • With local computer operation, a standard (non‐null) cable is used. • With a modem connection, a special "null modem" cable is required. For more information on the correct cable see Section 4. Accessing the command line using the serial port ...
Page 260 Data Bits – set to 8 • Parity – set to none • Stop Bits – set to “1” • Flow control – set to “none” Now click on “ok”. Once you have made the physical connections and programmed your terminal emulation software, you should see a blank terminal window. If all is ok you should see your own characters echoed back when you type. If not, check the settings and make sure you are properly connected. You may need to contact your software vendor for advice. To enter the system type: Login Telos <Enter> (if this does not work use Login <Enter>) You will see the following response: Logged in with user rights… >> Section 12: System Functions and Remote Control USER’S MANUAL...
Page 261 USER’S MANUAL : "help * <enter>” as shown below. To get a list of all available commands type >>Help * Built-in commands: Name...: printsender Parameter..: - Group...: Multicast Description: Show all defined Stream Senders. Name...: startsender Parameter..: [<Host name|IP address>] [<port>] Group...: Multicast Description: Start an stream sender host. Name...: addsender Parameter..: [<Host name|IP address>] [<port>] [<ttl>] Group...: Multicast Description: Add an stream sender host.
Page 262 USER’S MANUAL Description: Select analog/AESEBU input level in dB. Name...: zlog Parameter..: [SYS|ISDN|IMON|SEXT|TIME|ITIM] Group...: Zephyr-style Description: Switch on a combination of logging channels. No parameter will switch off logging. Name...: zloop Parameter..: <off|far|near> Group...: Zephyr-style Description: Loop mode far, near or off, Default off Name...: znetwork Parameter..: <ISDN|V35|Ethernet>...
Page 263 USER’S MANUAL Name...: etherrate Parameter..: <8|12|14|16|18|20|24|32|40|48|56|64|80|96|112|128> Group...: Stream Decoder Description: Choose bitrate for Ethernet operation. Name...: ipdchost Parameter..: [<host name or IP>] Group...: Stream Decoder Description: Set Decoder Host Name...: ? Parameter..: 1: <none> 2: <command> 3: <'*'> Group...: Misc Description: Display list of available command &...
Page 264 USER’S MANUAL with a setup index it is erased. If the exclamation mark is specified the autodial is activated. If only 'index' is specified this setup will be displayed. Name...: backlight Parameter..: <0..0x7F> Group...: Misc Description: Select the frontpanel LCD screen backlight. Name...: bitrate Parameter..: <56|64>...
Page 265 Description: Dump out the contents of the currently visible frontpanel screen. This is the UU encoded version of the file TELOS.BMP. When a FTP server host name or IP is specified the data will be uploaded to this server under the name \pub\Screen.uue.
Page 266 USER’S MANUAL Name...: ipftphost Parameter..: [<host name or IP>] Group...: FTP/Telnet Description: Set or display the server that is used to download system and loadset software updates. Default is ftp.zephyr.com. Name...: ipupdate Parameter..: [<file-name>] Group...: FTP/Telnet Description: Connect to FTP server name or IP address and download the specified system code file into the SIMM.
Page 267 Description: Select ISDN telco type. Unit must be reset in order for the change to take effect. Name...: levelin Parameter..: <cons|prof> Group...: Audio Description: Select consumer or professional level for analog inputs. Name...: levelout Parameter..: <cons|prof> Group...: Audio Description: Select consumer or professional level for analog outputs. Name...: location Parameter..: [<number0..29>...
Page 268 USER’S MANUAL Name...: micprocess Parameter..: [<mic1..4> <VoiceL|VoiceM|VoiceH|Music|Off>] Group...: Audio Description: Select the processing type for the specified microphone input. Name...: micpan Parameter..: [<mic1..4> <-10...+10>] Group...: Audio Description: Set the panning for the specified microphone input in case it is assigned to both input channels.
Page 269 reset statistics. Name...: pp Parameter..: ['in' [val]|'out' [val]] Group...: Misc Description: Override incoming or outgoing parallel port values. Return to normal operation if 'val' is not specified. Read current input & output value if no parameter is given. Name...: ppin Parameter..: [<bit1..8>...
Page 270 USER’S MANUAL Group...: Codec Description: Enter or display the V.35 highspeed encoder bitrate. Name...: v35deco Parameter..: [l2|l3|aac|aacld] Group...: Codec Description: Enter or display the V.35 highspeed decoder algorithm. Name...: v35enco Parameter..: [l2mono|l2stereo|l2joint|l3mono|l3stereo|l3joint|aacmono|aacstereo|aac joint|aacldmono|aacldstereo|aacldjoint] Group...: Codec Description: Enter or display the V.35 highspeed encoder algorithm. Name...: version Parameter..: 1: <none>...
Page 271 Open 123.345.678.901 <enter> (or select open from the file menu, depending on the Telnet client you are using). Substitute the IP address of your Xstream for the IP address shown. You can now login to the Xstream and use the command language as described above. Telnet is port 23. If you have difficulties with access through a firewall give this information to your network administrator. Some versions of Hyper Terminal support Telnet sessions as well. Or, you may be able to enter the following in your browser: Telenet:123.345.678.901 <enter> You will need to login in as described previously. Type: Login Telos <Enter> (if this does not work use Login <Enter>). NOTE: The Password can only be changed using the RS‐232 or Telnet command line. See section 12.2 for details. Control via Ethernet with Web Browser The Xstream can be controlled via a web browser. Of course the computer with the browser and the Xstream must be both connected to a TCP/IP network. Simply enter: http://109.876.543.210/ (substitute the IP address of your Xstream) on your browser. NOTE: The login procedure and web pages are different depending on whether you have the 10 Base‐T or 100 Base‐T (Livewire capable) hardware. Follow the corresponding instructions, below: 10 Base‐T units (original hardware) A login page will be displayed. For "password" type: (if this does not work “Telos <Enter>” “<Enter>”). 100 Base‐T units (later, Livewire capable units) You will need to login in. If requested for a user name, enter "user". By default, there is no password. Section 12: System Functions and Remote Control...
Page 272 The Xstream's HTML browser interface has been tested with the following browsers (others will probably work as well). • Firefox (Windows) 1.01 • Internet Explorer (Windows) versions: 5.00, 5.5, and 6.0 Zephyr Remote Large facilities where it is desirable to centrally locate equipment, may wish to explore the Zephyr Remote Software package from Software Authority. Using it you can control an Xstream via Ethernet or may control multiple Zephyrs and/or Xstreams via serial port. See www.softwareauthority.com to download the program for free. If you like it, you can purchase a key that allows unrestricted operation by contacting Software Authority. Panic Dial Panic Dial also allows simple dialing control of the Xstream via the parallel control port. See Section 11.7 (Detailed Menu Reference) for more information. Section 12: System Functions and Remote Control USER’S MANUAL...
Page 273: Deep Tech Information & Troubleshooting
13 DEEP TECH INFORMATION & TROUBLESHOOTING IMPORTANT! As with any piece of modern electronic gear, it is advisable that precautions be taken to prevent damage caused by power surges. Standard line surge protectors can be used to offer some degree of protection. It is the user’s responsibility to ensure protection adequate for their conditions is provided.
Page 274: Accessing The Hardware
13.1 Accessing the Hardware IMPORTANT! Always disconnect AC (mains) power and Telephone Circuit Before removing the Zephyr Xstream’s cover. Potentially lethal voltages are inside! Failure to observe the above warning may result in electric shock, injury or even death! Refer all servicing to qualified personnel.
Page 275 It is possible to shift the nominal inputs levels 4 dB lower (i.e. 4 dB more sensitive) therefore resulting in 0dBu(Professional)/‐15dBu(Consumer) levels. This is done by changing JP5 & JP6 on the audio board. See below for the possible settings. Early production units: JUMPER SETTING (JP5&6) Right Pins Left Pins Most units have the following jumpers: JUMPER SETTING (JP10 & 11) Pins 1&2 and 5&6 Pins 2&3 and 4&5 To change the input level operating range: 1. Remove the top cover as described above. 2. With the front of the Xstream facing you, look for the AES/EBU socket PC board protruding towards you from the rear panel. Below this board, at the bottom of the chassis, you will find jumper blocks JP5 & JP6 (early hardware revs) or JP10 & 11 (most units). Set the jumpers as desired (see tables above). 3. Replace the top cover as described above. 4. Set the Input Level Option in the AUDIO menu to “Consumer” or “Professional” as desired (see table, above). Assigning the XLR outputs to monitor mix 1 (MX and MXP version only) Newer model MX/P units allow the XLR outputs to be assigned to Local Mix 1 instead of Local ...
Page 276 3. Verify that the version number of this board is 1401‐00008‐006 (Rev F) or later. 4. Locate jumpers JP7 and JP8 on the MIC Audio Board. These are located near the left edge, about halfway back. 5. Move the jumper blocks from the Normal position (towards the back of the unit)) to the Monitor 1 position (towards the front of the unit). Enabling PS2 power on the ISDN S interface Early rev Xstreams have the ability to provide power on the ISDN S interface (to power a telephone for example). Pins 7 & 8 of the ISDN S interface on these units can supply ‐48 VDC to an external device such as a telephone. This can be disabled, which is the default setting. IMPORTANT TIP! Leave this power source disabled unless you know how you will be using it. Under no circumstances should this power be enabled if the "S" interface will be shared with a Zephyr Classic or ZephyrExpress.
Page 277: Updating Firmware
USER’S MANUAL troubleshoot. Therefore, we expect that most repairs will need to be done by Telos or our authorized representatives. See page III for details on obtaining service or support. However, there are a few sub‐assemblies, which are field replaceable. You can purchase the following spare parts, if desired. ISDN S + U Interface Telos Part # 1701‐00023 (old # 9456‐0000) for 10 Base‐T hardware only ISDN S + U Interface Telos Part # 1701‐00153 (for 100 Base‐T and 10 Base‐T hardware. Requires software version 3.x or later) V.35/X.21 Interface Option Card Telos Part # 1701‐00006 (old # 9454‐0000) Zephyr Xstream Power Supply Telos Part # 1281‐0006 Zephyr Xstream SIMM memory module (10 Base‐T hardware version only) Telos Part # 1166‐ 0005‐XS‐version (specify "Zephyr Xstream" and the desired version on your order) NOTE: When installing the 1701‐00023 ISDN S + U Interface be sure the following jumpers are correctly set: JP2 must short pins 1 and 2 (down position) JP1 and JP3 should be set as indicated above. 13.2 Updating Firmware The procedure for updating units is one area where early units (with 10 Base‐T Ethernet port) and newer Xstreams (with 100 Base‐T Ethernet and Livewire) differ substantially. The early unit request updates from the FTP server specified in the System menu (this might be a local machine running a special updater/FTP server, or it might be a remote FTP server such as the one at Telos. The newer units (with 100 Base‐T) are updated via the web browser. In this case, the appropriate files are downloaded from Telos to the local machine. Then, from the web browser, the user specifies the path to the files when the update process is initiated. Please consult the appropriate section for your unit. NOTE: You can determine if your unit has the newer 100 Base‐T hardware by checking your software version in the first SYSTEM screen. If your software version ends in "I" or "P" you have older 10 Base‐T hardware. If your version ends in "B" or "R" you have the 100 Base‐T hardware. 13.3 Updating Xstreams with the 100 Base-T hardware The Xstream has two internal memory “banks”. Each bank contains room for a complete ...
Page 278 This will disconnect any calls. Saving Bank 1 Software Software is always downloaded to bank 1 (the secondary bank). Downloading software (see below) will overwrite the software currently in this bank, if any. If you wish to save the software currently residing in bank 1 you can save it by moving it to bank 1 as follows: 1. Click on the Commit this version to Bank 0 box (see below). 2. Click on Apply. The node will now reboot. IMPORTANT! The node will reboot after you click Apply if you change the software version. This will result in loss of audio locally, and at any unit using the local sources. In addition you will reset your browser connection and will need to reconnect. Downloading new software A new software version can be downloaded into bank 1 as follows: 1. Go to the Telos web site: www.telos‐systems.com and download the desired software update to your computer (this should be the computer that you will use to access the node’s web page) by clicking on the link. Your local computer operating system should display a prompt to permit you to choose where you wish to locate the downloaded file. You can choose any convenient location, just be sure to note the drive and location where the file is to be saved. 2. Open a web browser and connect to the Xstream to be updated. Enter the complete path and file name for the software file (e.g. the file downloaded from the Telos site), or click on the Browse button to locate the file. Once the proper path and filename are displayed, click on Apply to download the file. The download process is lengthy. Consult the version notes for details. Section 13: Deep Tech Information and Troubleshooting USER’S MANUAL...
Page 279: Updating The Original (10 Base-T) Xstream
3. A successful download will be indicated by the new version being displayed in the Bank 1 field. If the download is unsuccessful the field for Bank 1 would be blank. 4. Click on the Bank 1 radio button and then Apply to reboot and run the new software version. IMPORTANT! The node will reboot after you click Apply when changing between software versions. This will result in loss of audio locally, and at any unit using the local sources. In addition you will reset your browser connection and will need to reconnect. 13.4 Updating the original (10 Base-T) Xstream Zephyr Xstream’s software is stored on a re‐programmable module. The Xstream can use FTP (File Transfer Protocol) over any IP network to download new firmware from an FTP server into the memory module. We advise you to contact Telos Customer Support (have your current software version and Serial number when you call) to discuss whether the latest version would better meet your needs. The Telos web site www.zephyr.com includes information on updating your Xstream. This information includes version notes describing the changes between the software version. There are two basic scenario’s. If your firewall allows FTP through, then the Xstream can download the new software directly from the Telos FTP site (ftp.telos‐systems.com). If your firewall does not allow direct FTP download, you will need to download our free tlsFTP server software and the Xstream software files. This download can be initiated at the Telos web site. You will need to download this file archive to an IBM PC compatible machine on the local network. The readme file included in the archive has detailed instructions on setting up the Telos FTP server. You will then use the IP address of your local FTP machine as the FTP site in the Zephyr Xstream’s system menu and then initiate the update process. If you are unsure of the IP address for the computer you are using for the update, go to a DOS window and type "ipconfig <enter>". To get to a DOS window go to the run window programs and type "command <enter>" (Win 3.1, 95, & 98) or "Cmd <enter>" (Win NT, 2000). Detailed update procedures follow. Updating 10 Base-T Xstreams from a local ftp updater.
Page 280 2. Start the Telos update server by double clicking. It is important that the download process completed without interruption. If local power is not reliable, you might wish to place the Zephyr Xstream (and local FTP server, if used) on an un‐interruptible power supply. 3. Connect the computer to be used for the update and the Zephyr Xstream to the same Ethernet network. The following diagram shows the usual connections. Note that a direct connection between the Xstream and the computer may be made, if the appropriate Ethernet “Crossover” cable is used. See Appendix 5 (Modular Cable Guide) for details on that cable, if needed. 4. Program the following information into your Zephyr Xstream. Ask your network administrator if you need help. 5. Press the <SYSTEM> button three times 6. Enter the following information in this menu: IP Address‐ The IP address for this unit. As with any computer on an IP network, the Xstream must have an IP address before it can be used over the network. SN Mask‐ The subnet mask is to determine the size of your local network. All packets addressed to a destination outside this local area are sent to the gateway node entered in the next selection. Normally you should enter 255.255.255.0 here. Press <SYSTEM> again and enter: FTP site ‐ This is the IP address of the computer running the Telos Update FTP server. 7. Update the system Section 13: Deep Tech Information and Troubleshooting USER’S MANUAL...
Page 281 Select FTP update… and press <SEL>. Press <5> or <6> to choose OK . Press <SEL>. The unit will reboot once the update is complete. It may also give a message about "invalid non volatile RAM values". This message is normal. 8. Press <SYSTEM> once and verify that the new software version is shown. If not, repeat the update. The update process is now complete. NOTE: It is not abnormal for the system to display a message "Some corrupt non volatile values were detected". You can disregard this message. Updating a 10 Base-T Xstream directly from the Telos ftp site Note: This method is preferred if your firewall allows access to the Telos ftp site (ftp.zephyr.com). In this case, you need only a connection to the internet ‐ a local computer is not needed. If your firewall allows FTP (usually port 21) through, then the Xstream can download the new software directly from the Telos FTP site (ftp.telos‐systems.com). If your firewall does not allow direct FTP download, see sections I and II, above. Software update step-by-step (Telos Remote Server) It is important that the download process completed without interruption. If local power is not reliable, you might wish to place the Zephyr Xstream on an uninterruptible power supply. 1. Connect the Zephyr Xstream to a network with internet access. 2. Program the following information into your Zephyr Xstream. Ask your network administrator if you need help. Press the <SYSTEM> button three times and enter the following information in this menu: ...
Page 282 DNS – This is the IP address of the DNS (Domain Name Server) you will be using. Press <SYSTEM> and enter: FTP site ‐ This would normally be ftp.zephyr.com if downloading from Telos FTP site. 3. Update the unit. Select FTP update... and press <SEL>. Press <5> or <6> to choose OK . Press <SEL>. The unit will reboot once the update is complete. It may also give a message about "invalid non volatile RAM values". This message is normal. 4. Press <SYSTEM> once and verify that the new software version is shown. If not, repeat the update. The update process is complete. NOTE: It is not abnormal for the system to display a message "Some corrupt non volatile values were detected". You can disregard this message. HOT TIP! If you need to "borrow" the IP address of a Windows computer to use when updating your Xstream, here are some tips: 1. Go to the MS-DOS command line of the Windows machine (Run " cmd <enter>" for Win XP, 2000 or Win NT;...
Page 283 HOT TIP! If you don't know a valid DNS (domain name sever) IP address, you can still update your Xstream from the Telos ftp site by obtaining the current IP address of the Telos FTP server as follows: 1. Go to the MS-DOS command line of the Windows machine (Run " cmd <enter>" for Win 2000 or Win NT;...
Page 284 • Locate the notch on the SIMM module to your right (towards the rear of the Xstream). • At a 30 degree angle from vertical, insert the SIMM module into its socket. It should slide in with no resistance. The module will be angled towards you. • Using your two thumbs push on the green area near either end of the SIMM module. The top of the SIMM module should pivot back and the catches on either side should click. If this does not happen remove the SIMM module (see step 3, above) and try again. 4) Reinstall top cover 5) You may need to reprogram some of your settings, check them after installing the new software. Please return the old SIMM module to Telos, via regular US Mail. Your prompt return of this module will allow us to continue to offer you upgrades in the future, at the lowest possible cost. Section 13: Deep Tech Information and Troubleshooting USER’S MANUAL...
Page 285: Troubleshooting
13.5 Troubleshooting The following sections discuss various areas where you may encounter challenges. We have categorized this section into logical problem areas to make it as painless as possible. Of course, you are always welcome to contact our support team if you get stuck! 13.6 General Troubleshooting Techniques Thinking about Problem Solving Despite best intentions, something could always go wrong. Sometimes troubleshooting a balky set‐up can make even the toughest engineer a Maalox and Rogaine addict. You can't fix any system without the right world view; a zeitgeist of suspicion tempered by trust in the laws of physics, curiosity dulled only by the determination to stay focused on a single problem, and a zealot's regard for the scientific method. Perhaps these are characteristics of all who successfully pursue the truth. In a world where we are surrounded by complexity, where we deal daily with equipment and systems only half‐understood, it seems wise to follow understanding by an iterative loop of focus, hypothesis, and experiment. The notions here apply whether you are solving problems at the system level or at the component level. At the system level, the actions you might take would be very different – checking cables, trying different menu settings – but the thinking is the same. Too many times, we fall in love with our suppositions. We are quick to overtly or subconsciously assume the problem being chased is due to lousy design, the stupid phone company, or the manager’s latest memo. Armed with a healthy skeptical attitude, the basic philosophy of troubleshooting any system is to follow these steps: • Observe the behavior to find the apparent problem; • Observe collateral behavior to gain as much information as possible about the problem; • Round up the usual suspects; ...
Page 286 Let us now cover each step of the troubleshooting sequence in detail. Step 1. Observe the behavior to find the apparent bug. In other words, determine the bug's symptoms. Remember always that many problems are subtle and exhibit themselves via a confusing set of symptoms. Step 2. Observe collateral behavior to gain as much information as possible about the problem. Does the LCD's problem correlate to an LED flashing? Try to avoid studying a problem in isolation, but at the same time be wary of trying to fix too many at the same time. No one is smart enough to deal with multiple problems all at once – unless they are all manifestations of something more fundamental. Understand that if a problem relates to telephone network routing, that your results may vary call to call. If you suspect this may be the case, you should make repeated calls (as many as 15!) and note the success rate. You can then change something and retest to see if the success rate was affected by your experiment. Network related problems will frequently vary based on which party places the call, the long distance carrier, and the bit rate; you can try varying these. Step 3. Round up the usual suspects. At the system level, always suspect the menu set‐up, the cables, the Phone Company line setup, the punch‐blocks, etc. At the component level, many computer problems stem from the same sources. Never, never, never forget to check Vcc! Step 4. Generate a hypothesis. Before changing things, formulate a hypothesis about the cause of the problem. You probably don't have the information to do this without gathering more data. Sometimes you will have no clue what the problem might be. Sometimes, when the pangs of desperation set in, it's worthwhile to try anything practically at random. You might find a bad plug, an unconnected line, or something unexpected. Look around, but be always on the prowl for a working hypothesis. Step 5. Generate an experiment to test the hypothesis. Change the ISDN connection to a known good line; call known good phone or hybrid at the other end; if long‐distance doesn’t work, try a local call. NOTE: You should plan your tests to eliminate 50% of the possible problems in one test, if possible. Just keep careful track so you know what you have eliminated. Step 6. Fix the problem. A Final Thought… Constantly apply sanity checks. Many years ago the Firesign Theater put out an album called "Everything You Know is Wrong". Use that as your guiding philosophy in troubleshooting a Zephyr Xstream. For example, just because you checked the Telco line last night, and it was fine, does not mean that it's OK now. At 3:00 AM when the problems seem intractable and you are ready to give up engineering, remember that the system has worked and will work. Never panic–you are smarter than it is. Section 13: Deep Tech Information and Troubleshooting USER’S MANUAL...
Page 287 USER’S MANUAL 13.6.1 Diagnostic Indicators The Xstream has a number of diagnostic indicators to help in troubleshooting difficulties. They are: SYNC status LED When ISDN is selected for the Interface (in the 3 TEL menu), the SYNC LED (above the SEND bargraph) should illuminate whenever power is applied. It also indicates that the Zephyr Xstream has basic functionality. If not, the Xstream has not completed its boot cycle, or major clock problems exist within the unit. When V.35 is selected for the Interface (in the 3 TEL menu) the SYNC LED indicates that a clock is present at one of the V.35/X.21 digital ports, and it is at the correct frequency to match the Bitrate (as selected in the first CODEC menu). Since this clock is passed through from the CSU/DSU, it is the best indication that the CSU/DSU is connected and has at least basic functionality. When in “Ethernet” mode for Interface in the CODEC menu the sync light will generally flash periodically for about 5 minutes after connection is established. This is due to the fact that with Ethernet there is no master clock source as there is in V.35 or ISDN mode, so one of the units must lock itself to the clock of the other unit. To do this without side effects the change is made very gradually and until the process is complete the SYNC LED may flash periodically. RECEIVE Lock LEDs The LOCK LED illuminates to indicate that the decoder is receiving valid data. When a Receive mode capable of independent operation using the two B channels (i.e. L3 Dual/Mono or G.722) is used, the LOCK 1 LED indicates valid data for “Line 1” while the LOCK 2 indicates this information for “Line 2”. When a Receive mode is used that requires 112/128 kbps of data both LEDs illuminate once the correct pair of data‐streams are detected. Both LEDs will also illuminate if a valid data stream is received in V.35 Single mode. If this is flashing, or off, there is either an incorrect setting of the Receive mode, Sample Rate, or Bit Rate (these parameters are all set in the second CODEC menu). Or, there is corrupted data being received. If incoming the data suffers from bit errors, the audio will be briefly muted and the appropriate LOCK LED will extinguish until valid data is received again. When using Ethernet mode and receiving AAC, the lock light will flicker off when bad data is ...
Page 288 The ISDN S interface is generally used outside of the USA & Canada. If you are not using the ISDN U (2-wire) interface you can disregard this LED. In that case, this LED will be flickering. FLASHING A slow flash means the NT1 has completed initializing the U interface (the line side) and is initializing the S interface to the rest of the Zephyr Xstream. If this state persists for more than 40 seconds, it is most likely a hardware problem with the Zephyr Xstream. You should “reseat” the ISDN card. Check for foreign voltage on the ISDN (you should see about 48 volts DC [possibly pulsing] on the line) If the problem persists, contact Telos Customer Support. OFF This is an illegal state and should not occur. In rare cases this could be the ISDN circuit. If disconnecting the ISDN line causes the LED to begin flickering, suspect this. Otherwise, if the Zephyr Xstream seems to be booting up, it indicates a bad Terminal Adaptor Card (90%). You should “reseat” the ISDN card. If the problem persists, contact Telos Customer Support. ** ALTERNATING If the state alternates between more than one state, and continues to cycle between those ...
Page 289 ISDN card. This state could indicate a marginal ISDN line. If the problem persists, contact Telos Customer Support. * NOTES: ** If you reboot the unit and the LED stays either off or on, even with no ISDN attached, there is a problem. If the Zephyr otherwise seems to be operational (i.e. the keyboard and LCD screen and meters are functioning) it indicates a bad Terminal Adaptor Card (95%). You should “reseat” the ISDN card. If the problem persists, contact Telos Customer Support. * If the phone company tests the line and claims it is ok, ask them what brand and model of “switch” you are on. Also, ask them if you are on “straight copper” or if you are on any kind of “repeater” or “loop extender”. Ask them how far you are from the switch or nearest repeater. Call Telos Customer Support and give them this information. External NT1’s have one or more indicators, which can help differentiate line vs. Zephyr Xstream problems. See your NT1 manual for details. The TEL Status Screen The status of each B channel is shown here. Press the TEL button once to view this screen. The same status information is displayed on the default timeout screen. The system tries to initialize the ISDN circuit upon boot up. ISDN circuits requiring SPIDS (generally needed in the USA & Canada, never needed outside these countries) will display SPID Error if the SPID for specific channel is incorrect. Note that you may get a false Ready indication if the TELCO setting (in the second TEL menu) is set incorrectly or if no SPIDs are entered when they are required. For status of the NT1 observe the state of the Green LED next to the U interface. Available line status messages are: Inactive SPID Pending SPID Error Ready Pending Seize Section 13: Deep Tech Information and Troubleshooting USER’S MANUAL...
Page 290 Call Proceeding Outgoing Ring Accepting Call Connected XXX Disconnecting Ended Ring 13.6.2 Using Loop Modes for Troubleshooting Loop Mode This item allows the Zephyr Xstream to loop audio through the system. • None This is the normal operating mode. None of the loopback paths are turned on. Studio Input is sent as send audio, and caller audio is presented at Studio Out. • V.35 This loops the incoming V.35/X.21 Receive data on each V.35/X.21 port back out the Xmit data pins on the same port. NOTE: a valid external V.35 clock is required. • Audio This loops the AES/EBU input (and analog signal inputs after Analog‐to‐ Digital conversion) back to the AES/EBU output (and analog output via the Digital‐to‐Analog converter). With this mode engaged you can make signal to noise, THD, and frequency response measurements through the system to check the input and output stages. Section 13: Deep Tech Information and Troubleshooting USER’S MANUAL Outgoing call in process; Telco switch has acknowledged the ...
Page 291 USER’S MANUAL • ISDN This option causes two different things to happen. a) Data coming in the ISDN interface (when INTERFACE is set to ISDN ) is sent back out the same channel(s) of the interface. Therefore, the unit on the other end should be able to receive its own audio back (assuming it is configured for compatible Transmit and Receive modes). If this is the case, the network path is functioning. b) Codec data from the encoder is passed back to the decoder (when INTERFACE is set to ISDN). Therefore, you should receive your own audio back (assuming it is configured for compatible Transmit and Receive modes) with delay and quality dependant on the coding mode used. This tests over 70% of the Xstream's functionality. This mode can be used to audition the various coding modes even when no ISDN circuit is present. AUDIO LOOPBACK SWITCHES ENCODER SEND audio DECODER audio Xstream The AUDIO and ISDN loopback modes are illustrated above. The switches on the left side are activated for the AUDIO LOOPBACK mode while the switches on the right are activated for the ISDN LOOPBACK mode. Section 13: Deep Tech Information and Troubleshooting ISDN LOOPBACK SWITCHES data...
Page 292 13.6.3 Audio Difficulties Condition: ISDN Connecting OK, But No Audio What is the state of the LOCK LEDs? Check the LOCK LED. On means OK, connected and receiving valid data. If not illuminated, try cycling through receive modes to see if you can “find” the mode at the other end. What happens when you call yourself? Try calling yourself, from one ISDN line to the other (assuming you have both available). Use a 56 kbps rate first, as this is the most universal. Set Rcv mode to L3 Stereo and Xmt to L3 Dual or L3 Stereo) Audio should work in both directions. This confirms that the Zephyr is OK, and that the problem lies somewhere downstream, probably in the Telco’s connection facilities either between its own central office switching equipment or to the long‐distance carrier you are trying to use. What happens to your local audio when you try an ISDN loopback? If successful, this eliminates over 70% of the Xstream circuitry as the culprit. Remaining suspects would be primarily the ISDN card, or the V.35 card (if relevant to your problem). Remember, Xmt and Rcv modes must be compatible. Are the coding modes set properly? The Xmt (transmit) and Rcv (receive) modes and the bitrate must be set in conjunction with those of the unit at the other end. The Bit Rate and Sample Rate setting must also match. What happens if you try a different long distance carrier for the call? What happens if you try the call at 56 vs 64 (or vice versa)? What happens if you try reversing the direction of the call? What happens if you try placing a local call as a long distance call? Since most of the network uses separate transmit and receive paths, it is entirely possible to have a path in one direction but not the other. The above tests may force you through the network via a different path. If local calls are ok, but long distance calls fail with all three major long distance carriers, then it is likely there is a trunk problem between your local CO and the tandem switch. Condition: No Audio In Both Directions Does the Zephyr Xstream’s loopback isolate the problem? ...
Page 293 Are the Xstream‘s modes set properly? The Xstream’s Transmit Mode and Receive Mode (both located in the second CODEC menu) must be set to the proper conditions at each end in order for the system to work. See Section 10 (Application Guide) and Section 11 (Detailed Menu Reference) for the correct settings. Sometimes cycling between modes can clear a decode problem. Is audio properly connected and configured? Are the Zephyr Xstream’s front panel Send bargraph meters indicating signal present at a correct level? If not, check audio connections and cables. Is the Input level setting in the AUDIO menu set correctly? Condition: Audio In One Direction Only This problem can be either analog or digital in nature. Unlike analog telephone lines, the two directions of a digital line are mostly independent. It is possible to have a telephone channel failure in one direction only. One of the following may serve to force you through the network via a different path. What happens if you try a different long distance carrier for the call? If you try the call at 56 vs 64 (or vice versa)? Try reversing the direction of the call? Try placing a local call as a long distance call? Are audio signals and connections OK? It goes without saying – it’s got to be plugged‐in to work. If the problem is only with one channel reversing A and B audio connections will help problem, isolation. Do you have a good telephone network path in both directions? Isolate the problem area by using the loopback techniques described above. Remember, if unit can lock to itself via the far end unit's loopback, then the network path is working. Condition: Audio Distorted Mixer models ‐ Is the AUDIO menu Ch "x" gain selection set correctly? Applying a line‐level signal to a system configured for microphone levels will make for some pretty awful distortion. Check the clip indicators on the first AUDIO menu. Check the setting of the Proc X in AUDIO menus 2‐5. The Voice Medium and Voice High settings can be very aggressive. Non Mixer models ‐Is the AUDIO menu "Level In" selection set correctly? Are jumpers JP5 and JP6 set ...
Page 294 Check the Send bargraph meter for proper indication. Confirm that the top Send LED is flashing not at all. Input/send gain is set on the AUDIO menu screen. Output setting correct? The Level Out setting in the AUDIO menu should match to the equipment that accepts the Xstream's audio feed. 13.6.4 ISDN Difficulties Ready” Condition: Can’t Get ISDN “ First try disconnecting the ISDN circuit(s) in question and waiting a full 60 seconds (leave the Zephyr Xstream on) and then reconnect the circuit. Are you connected to the correct jack on the rear of the Zephyr Xstream? If in the USA & Canada: you will probably be using the lower (RJ‐11 style) jack. Outside the USA & Canada: you will probably be using the upper (RJ‐45 style) jack. Try dialing. Some Euro ISDN lines will show inactive until they are in use. Are the jumpers correct on the ISDN Interface card? JP2, if present, must be jumpered (newer cards no longer have this jumper). What kind of ISDN do you have, and is the Xstream set to match it? The telephone company should have given you the ISDN protocol when the line was ordered. If you find yourself in the field without this information, it is usually possible to piece together what you need. • If you are outside the USA & Canada, you most likely have ETS 300 (Euro ISDN). If you are in Japan, you have INS64. These users do not need to enter SPIDs. MSNs are optional. • If you are given SPIDs, you probably have National ISDN 1 (NI‐1) or DMS 100 (DMS‐ Cust.) for your ISDN protocol. One SPID means one channel was activated, rather than the usual two. You won’t be able to make a stereo connection. National ISDN 2 is backward compatible with the Zephyr Xstream (use Natl.I‐1 for Telco in the third TEL menu) • No SPIDs and only one phone number generally means AT&T Custom (PTP) protocol. ...
Page 295 You may be able to guess the correct Telco setting based on other available information: • If you were given only a single phone number and no SPIDs try AT&T Cust (PTP) • If you were given 2 SPIDs with the format: 01+7 digit phone #+0 try AT&T Cust (PTP) • If you were given 2 SPIDs with the format: 01+7 digit phone #+000 or 01+7 digits +011 try Natl.I‐1 • If you were given 2 SPIDs with the format: area code+7 digit phone #+4 digits (for example 0101, 0100, 0111, or 1111) try Natl.I‐1. If that fails Try DMS Cust • If you were given 2 SPIDs with the format: area code+7 digit phone #+1 digit (for example 1, or 0) try DMS Cust. If this fails try Natl.I‐1 • If you were given 2 SPIDs with the format: area code+7digit phone #+2 digits (for example 01, 11, 02, or 00) try DMS Cust. If that fails try Natl.I‐1 • If you were given 2 SPIDs with the format: area code+7digit phone #+3 digits (for example 000,001) try Natl.I‐1. If that fails try DMS‐Cust • If you were given a single phone number and a single SPID with the format: of 01+7 digit phone#+0 try AT&T Cust (Do not enter this SPID) • If you were given a single phone number and a single SPID with the format: of 01+7 digit phone#+00 try Natl.I‐1. If that fails try AT&T Cust • If you were given a single phone number and a single SPID with the format: of area code+7digit phone#+4digits (for example 0101, 0100, 0111, or 1111) Try Natl.I‐1. If that fails try DMS Cust ISDN TIP! TIPS ON SPIDS 1. Incorrectly entered SPIDs are the most common problem USA/Canada users have placing ISDN calls.
Page 296 ISDN TIP FOR BELLSOUTH CUSTOMERS! Despite the fact the BellSouth originated the proposal for the standardised SPID (see above) many regional Bell South offices leave off the last two digits of the SPIDs. If you are a BellSouth customers and are given a SPID of the format: Area Code + 7 digits + 01 and you get "SPID Error"...
Page 297 SPID Error ‐ This option is only valid when the Telco selection of the TEL menu is set to “Natl.I‐1” or “DMS Cust.”. The SPID entered for the indicated channel is has not been accepted by the Telco switch. You may have entered it wrong, or the Telco may have given you the wrong SPID (or entered the SPID incorrectly in their switch) or ISND protocol type. Ready – This indicates that an ISDN S interface is active (i.e. external NT1 is present or internal NT1 has completed startup). It indicates that all handshaking with the central office is complete (the SPIDs, if required and entered, are correct) and the line is ready to be used. Connected: xxx ‐ The channel is connected to "xxx". "xxx" is the caller ID of the calling party, if available for calls originated from the far end. "xxx" is the number dialed for calls originated by the Zephyr Xstream. A call timer will be displayed on indicating the length of time the call has been connected. Seize Pending ‐ Outgoing call in progress; Request for call has been sent to the Telco switch and Zephyr Xstream is awaiting acknowledgment. Seized ‐ Outgoing call in progress. Waiting for local switch to acknowlege. Proceeding ‐ Outgoing call in progress. Telco switch has acknowledged the call request and is awaiting routing information from the network. Outgoing Ring ‐ Outgoing call in progress; far end has acknowledged the call and has generated a call alert signal (i.e. it is "ringing"). Ring: xxx ‐ Incoming call from "xxx" has been detected and is being processed. "xxx" is the caller ID of the calling party, if available. If auto ans. (in the TEL menu) is set appropriately the system will answer the call momentarily. Accepting Call ‐ Zephyr Xstream has accepted an incoming call and is waiting for the Telco switch to connect it. Disconnecting ‐ Request to disconnect has been made. Awaiting acknowledgement. Ended ‐ Request to disconnect has been made and the call has been cleared. Two caveats regarding the Ready indication: If you have this indication with the Telco set to AT&T Cust. (PTP) and you cannot dial, it is likely that your line is Natl. I‐1 or DMS‐100. The AT&T custom protocol does not require SPIDs, so the Zephyr Xstream does not send or check them; it just assumes the line is OK in this regard, so it gives the Ready indication, though the line is waiting for the SPIDs. Also, be sure that the SPID 1 and SPID 2 fields (in the second TEL menu) are blank when using AT&T Cust. (PTP). You will also get see a false Ready indication if no SPIDs are entered and your Telco is set to Natl I‐1 or DMS Cust. IMPORTANT! It is normal for ETS300 lines to go to Inactive when not used for a time. If the system can dial normally under these circumstances, this indication is not a cause for alarm.
Page 298 Is the problem only with one BRI circuit? Does the problem move when you switch the line to another ISDN unit? If the problem remains with the line, then call your Telco. If you are supposed to be using the Xstream's built‐in NT1 (i.e. the U interface) be sure to note if you have U Sync (note the green LED near the U jack on the rear of the Xstream). If you have a fast blinking U Sync light the problem is most likely your ISDN circuit. If the U Sync LED is lit solidly, then suspect a line configuration (“provisioning” in Telco‐speak) problem (also review the information in the previous questions). Are you using the Xstream’s internal NT1 ? (USA & Canada only) The LED indicator of the status of the ISDN connection is on the rear panel near the U jack. If the circuit is inactive, the LED will remain in the rapidly flashing state. This state indicates a loss of the ISDN BRI circuit at the lowest (physical) level. Try another line cord or try another line in to see if the same thing happens. If the NT1 can contact the central office, the LED will blink slowly (about 1 Hz). If the LED remains in this state there is an internal Xstream problem or you could be connected to the wrong jack on the back (Users in USA & Canada should be using the lower, RJ11 style, jack). The LED will come on solidly when all handshaking is completed and the basic (low level) line connection is good. If the LED stays on, with the ISDN disconnected, or does not come on at all, there is a hardware problem. See above for more on this LED. Condition: Line 1 will accept call however Line 2 will not Have you entered Directory numbers in the MSN/DN1 &2 entries in the TEL menu? These are required if you are using DMS‐Custom or Natl I‐1 for your ISDN type. Condition: Can place calls, or receive calls, but not both If your line is using National ISDN and your Telco is set to DMS Cust., or vice versa, you may see ...
Page 299 Try a call to another local number. Have the far end try this test as well. Try calling yourself, both using both “call type – Zephyr” and “call type – phone”. Your line may be configured for only voice (CSV) or only data (CSD). If you cannot complete this call try dialing with a 9, 8, 7, or 6 before the number. Sometimes ISDN lines are set up as Centrex service even if you did not request this. Note the Cause Code or Words displayed when the Zephyr Xstream releases the line. See Appendix 3. Is your long‐distance carrier OK? If you can successfully call locally, but long‐distance calls don’t go through, the Zephyr Xstream usually displays Far end disconnect, or No route available as the ISDN “Cause” phrase. You may try a different by dialing the appropriate 10XXXXX prefix (in the USA & Canada only). Some codes we have found that are worth trying are: 1010222 (MCI) 1010288 (AT&T) 1010333 (Sprint) You will dial as follows (dial prefix, if required) + 10xxxxx + 1 + area code + 7 digit phone number. In the USA, you can determine who your long distance carrier is by dialing 1+700+555+4141 using the PHONE call type. If Intra‐LATA long distance competition has reached your area you can determine your Intra‐LATA long distance carrier by dialing 1+area code+700+4141. What does the ISDN “Cause” phrase say? This phrase, which will appears on the ISDN status screen whenever the line is released, comes directly from the telephone company equipment, and can be valuable for troubleshooting. Generally, these phrases are self‐explanatory. Getting them means you are at least talking to the Central Office equipment. There is a full description of the phrases and their meaning in Appendix 3. Condition: Intermittent ISDN Problems These are among the most difficult kinds of problems to solve. Rest assured, however, that we have yet to lose one! Does the problem occur on all of the ISDN BRI circuits or just a single BRI circuit? If your Zephyr Xstream works fine on another circuit, then it is unlikely it is the cause of your troubles. Tell the Telco this, and give them the phone numbers for the good line. If the problem is limited to a single BRI circuit (i.e. both “lines” on that BRI are showing the problem) then the problem is probably with that BRI circuit. Section 13: Deep Tech Information and Troubleshooting USER’S MANUAL...
Page 300 Assume nothing, and suspect everything. Leave no stone unturned while searching for the answer. Any other ISDN troubleshooting Tips? An important tool for understanding fundamental compatibility problems is the Zephyr Xstream’s integrated ISDN analyzer. This allows examination of the transactions between the Telco line and the Xstream, so that a judgment can be made as to whether the problem lies within the Xstream or with the programming of the Telco switch. This can be useful when the line itself tests good, but something is incorrectly configured at the Telco. The usual cause of inability to dial problems is a Telco problem, such mis‐programming. If you are convinced that the problem lies with the ISDN line configuration, you may be able to convince the Telco to have an engineer look at the line with an analyzer. This works much like the Zephyr Xstream’s internal analyzer, but with more detail, and in a format that is more readily understandable by telephone people. This procedure may sometimes be the only way to get attention to a line configuration problem. It is hard to deny evidence provided by their own gear. In rare cases, there could be a software bug in the Zephyr Xstream that reacts badly to something coming from the ISDN line. Telos customer support will need the details about your lines, Zephyr Xstream version and configuration. They will probably need you to use the protocol analyzer and a PC to make a log of the problem and will then communicate this information to our engineering folks for analysis. It is critical that you give them a precise description of exactly what happens when the problem occurs, and at what point in the log we should focus our efforts (i.e. the problem happened 45 minutes after we began this log). We are happy to help, but we must rely on your eyes and ears to help you. See Section 12 (System Functions and Remote Control) for information on connecting to the Xstream’s serial or Ethernet port and using the command line interpreter to turn on ISDN logging. The most useful logs to enable for this purpose are “ISDN”, il2, il3, and iie. Section 13: Deep Tech Information and Troubleshooting USER’S MANUAL...
Page 301 13.6.5 Ethernet Related Difficulties Condition: Unable to Telnet into the Xstream Is the Green link LED on the Ethernet jack lit? Is the "Link" LED on the hub or other equipment lit? If there is no link light, the Ethernet link is not up. The most likely cause is a bad cable. Other possible causes are a possible failure of the Xstream's Ethernet port or a problem with the hub (or other device at the far end). If you are connecting directly without an Ethernet hub, note that a special cable must be used (See Appendix 9 ‐ Modular Cable Reference) The Xstream’s command line interface is on the default Telnet port (Port 23). Verify your Telnet client is using this port. Does the Amber activity LED on the Ethernet illuminate occasionally? Whenever there is an active Ethernet link, you should see occasional activity of this LED. Lack of activity may indicate a problem with the Xstream. Reboot the Xstream and re‐ check Does the Xstream have an IP address entered? Does the Telnet computer have an IP address? Are they in the same IP range? The Xstream uses IP (Internet Protocol). IP addresses are required. Each device on the network must have unique IP address. Can you "Ping" the Xstream from the Telnet computer? If you cannot "Ping" the Xstream there is most definitely a problem. Make sure the Xstream's IP address is entered correctly. Try to "Ping" some other computer from the Telnet computer to isolate the problem. If you can "Ping" the Xstream, but you cannot Telnet into it, there may be a firewall between the two devices. Your network administrator should be able to open the firewall for Telnet access between these two IP addresses. (Note Telnet is port 23). Condition: Unable to connect to a distant site using IP (i.e. to receive audio) Can you "Ping" or "Telnet" to the distant Xstream from another computer on your local network? ...
Page 302 Condition: Able to connect and lock to a distant site using IP, but audio dropouts occur frequently Do local LAN connections work ok (or with a greatly reduced number of dropouts? Increasing the buffer setting may help. Also note that the Xstream has error concealment technology when using AAC, so we recommend using that protocol for this reason. Try increasing the buffer size in the CODEC menu What QOS (Quality of Service) mechanism is in place on the IP path between the two sites? Such QOS control is currently impossible if the path includes the public Internet. QOS options on a private network are limited. If no other equipment is sharing the IP link, then QOS is a non‐issue. As soon as other equipment shares the link, it has the ability to use all bandwidth at times it needs it (when downloading or transferring a file, for example). Your network equipment and IP link service provider may be able to assist with strategies. A qualified Information Technologist will be invaluable. One crude strategy, when only a few devices will share the link, is to use the Ethernet port to limit bandwidth. For example, if your primary point‐to‐point link is at 100 Mbps (mega bits per second), and it is fed from a single Ethernet hub using an Ethernet 100 Base‐T (100 megabit) hub you could connect a limited number of 10 Base‐T (10 megabit) devices. In this case you should be able to use eight to nine 10 Base‐T devices. These devices must not use 100 Base‐T or they can swamp the 100 megabit connection. The same setup could be used with smaller number of 10 Base‐T connections if the link were at a lower rate (i.e. you could connect three 10 Base‐T devices if the link were 50 Mbps). Unfortunately, the usual case is that the point‐to‐point link is at a rate much lower than 10 mbps ‐ often, 2.048 or 1.544 kbps (kilobits per second) or less. In this case, the ...
Page 303 Condition: Unable to connect when using SIP mode. If standard streaming works, and SIP streaming does not, then the proper port for SIP is probably not been opened up on the firewall. Both the SIP (default is 5060) and the UDP (default is 9150) ports must be available when SIP is used. Audio in only one direction can be caused by having the UDP port open only on one end. 13.6.6 Other Difficulties Condition: Xstream powers up, but the writing on the LCD is too faint to read. Possibly someone changed your contrast setting. We made this the first selection in the <SYSTEM> menu for a reason. Follow these steps: 1. Press the <TEL> button once. 2. Press the <SYSTEM> button twice. 3. Press the <SEL> button once. 4. Press the <5> button repeatedly until the screen contrast is at the desired level. Press the <SEL> button to save your setting. ...
Page 304 For continued protection against fire, replace fuse only with same type and value. If fuse replacement is required, please note: For continued protection against fire, replace fuse only with same type and value. See the section Gaining Access for information and important cautions. Does the Xstream have any functionality? Upon initial power‐up, the LCD shows "crosshairs", then briefly a black screen. Finally the version and the Zephyr Xstream logo. During the initializing period, a self‐test is performed. If there is a problem, this is reported on the LCD screen and progress is halted. The entire message should be noted for discussion with Telos customer support. If the failure is does not involve a critical portion of the Xstream, you may be given the option to "continue". We give this option for emergency desperation use only ‐ you will need to determine if you unit will work for the intended purpose. You should have the machine serviced as soon as possible if this happens. If the unit does not reach the show the Zephyr Xstream logo and software version, suspect a problem with the system processor, system clocks, or memory. If the "Crosshairs" are showing on the LCD screen, it may mean that the front panel and the main board have lost communication ‐ possibly because the main board processor rebooted for some reason. Or, possibly because the main board did not boot up. In the former case, the unit may be able to accept calls and connect without front panel ...
Page 305: Specifications & Warranty
14 SPECIFICATIONS & WARRANTY 14.1 Specifications General • Full duplex, high‐fidelity codec using MPEG‐2 AAC, MPEG‐4 AAC‐LD MPEG‐2 Layer‐3, MPEG‐2 Layer‐2, AACPlus, and G.722; fully compliant with international standards. Frequency Response (+0/‐1dB, swept sine tone procedure) • AAC all modes except Stereo 64: 20‐19,800Hz at 48kHz fs., 20‐15,000Hz at 32kHz fs. • AAC Stereo 64: 20‐10,000Hz at 48kHz fs., 20‐7,000Hz at 32kHz fs • AACPlus mono (for use reception from the Xport): 20 –15,000 Hz 48kHz fs • Layer‐3 all modes: 20‐16,000Hz at 48kHz fs., 20‐15,000Hz at 32kHz fs • Layer‐2 mono, dual‐mono: 20 Hz‐7.8kHz/9.8kHz • Layer‐2 mono 20 Hz‐ 8.6 kHz at 24 kHz fs. • Layer‐2 joint stereo: 20‐20,000Hz at 48kHz fs. 20‐15 kHz at 32kHz fs • G.722: 20‐7,500Hz. THD+N • Audio loopback, 48kHz fs, analog I/O, input at 1kHz +20dBu: 0.004% Dynamic Range • A Weighting, AAC, Layer‐3 or 2 end‐to‐end: 101dB typical Send Input Active balanced with RF protection. ...
Page 306 Level: Front panel selectable for ‐10 or +4dBu, nominal. • Impedance: < 33 (x2) • XLR male AES/EBU Digital I/O (non‐mixer version only) • Sample rates supported: 32, 44.1 and 48kHz • Rate conversion: Input and output independently selectable. Can be bypassed. • Input clock: From external source or Telco clock. • Output clock: From transmission sample rate, external source, or AES/EBU input. Inverse Multiplex/Demultiplex Internal channel splitting/combining of two network channels for stereo modes. • AAC: Telos Zephyr™ protocol. • AAC‐LD: Telos Zephyr™ protocol. • Layer‐3: FHG/Telos Zephyr™ (Buchta) protocol. • Layer‐2: CCS CDQ™ protocol compatible. Optional V.35/X.21 Direct Digital Interface • Two ports, both V.35/X.21. Automatically selected when the appropriate cable is connected. ISDN Interface • Compatible with National ISDN, AT&T 5ESS custom, Northern Telecom DMS‐100 custom, Siemens EWSD, INS 64(Japan) and EURO‐ISDN (ETS‐300). Compatibility and approval pending in some countries; contact Telos for current status. LAN Interface ...
Page 307 IP Support • via Ethernet 100 Base‐T Ethernet port. • Uses fixed IP address • Port 23 supports Telnet sessions • Port 80 used for HTML browser interface • Port 8080 supports MPEG streaming in MPEG Layer 3 or AAC • Port 5060 for SIP signaling • Port 21 used for FTP software updates (10 Base‐T units only). Software Upgrades • Downloadable from Telos; browser based upload via 100Base‐T Ethernet connection. Control Ports • Eight bi‐directional inputs/outputs for end‐to‐end contact closure emulation. • Inputs: Open collector, closure to ground. Integral pull up. Supports up to 24 volts DC inputs • Outputs: Sink up to 125 mA to ground. Integral pull up to 5 volts (voltages greater than 5 volts may be used with external pull‐ups) Resolution • Send Input: MPEG 24 bits; G.722 16 bits • Receive Output: MPEG 24 bits; G.722 16 bits Power Supply •...
Page 308: Zephyr Xstream Limited Warranty
14.2 Zephyr Xstream Limited Warranty This Warranty covers “the Products,” which are defined as the various audio equipment, parts, software and accessories manufactured, sold and/or distributed by TLS Corp., d/b/a Telos Systems (hereinafter “Telos Systems”). With the exception of software‐only items, the Products are warranted to be free from defects in material and workmanship for a period of one year from the date of receipt by the end‐user. Software‐only items are warranted to be free from defects in material and workmanship for a period of 90 days from the date of receipt by the end‐user. This warranty is void if the Product is subject to Acts of God, including (without limitation) lightning; improper installation or misuse, including (without limitation) the failure to use telephone and power line surge protection devices; accident; neglect or damage. EXCEPT FOR THE ABOVE-STATED WARRANTY, TELOS SYSTEMS MAKES NO WARRANTIES, EXPRESS OR IMPLIED (INCLUDING IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE).
Page 309: Appendices
CCS/Musicam • CDQ 1000 • CDQ 2000 • CDQ Prima • Roadrunner Comrex • Nexus • DXP • DXR Dialog 4 • VP Pro • Slimline (must set "compatibility" in Codec menu to "Slimline") Telos • Zephyr • ZephyrExpress • Zephyr Xport (over POTS or ISDN) We continue to confirm interoperability with other codecs. Appendix 1 – Codec Interoperability Information USER’S MANUAL...
Page 310 USER’S MANUAL Appendix 1 – Codec Interoperability Information...
Page 311: Appendix 2: Finding Public Codec User Lists
Appendix 2: Finding Public Codec User Lists When you need to find a Zephyr Xstream, ZephyrExpress, Zephyr, or compatible codec to communicate with in another city, there are a number of resources to which you can turn. There are thousands of sites around the country – and the world – which offer a wide variety of services. Some are music recording studios; some are broadcast stations; some are recording studios specializing in broadcast projects; and many are home studios belonging to announcers or producers. Some come with award‐winning engineering or vocal talent, while others are offered “bare.” Chances are you’ll find what you are looking for somewhere below. Telos World Wide Web site http://www.telos‐systems.com will connect you to a variety of information about Zephyr Xstream, Zephyr ZIP Codec, Zephyr iPort, and ISDN, including site pointers. Audiobahn A list of broadcasters and sound studios using codecs, maintained by Jay Rose’s Digital Playroom, a broadcast promo studio in Boston. These sites have submitted their names so other professionals can call them for news feeds help with remotes, audio transfers, and so on. All services are optional, and fees are negotiated by the parties involved. The list is maintained as a public service to help you find each other... after that, it’s up to you. Audiobahn is updated approximately once a month. Submission information is on the list, and has to follow a precise format. Comments may be sent by e‐mail to: jay@dplay.com. http://www.dplay.com Digifon List Dave Immer’s Digifon publishes a list of codec users around the world, heavily geared toward recording studios. Most (those using equipment which supports MPEG Layers II or III) are able to work with ZephyrXstream. Those that cannot (mostly APT‐X, but some Dolby) are able to do so via various bridge services, including one offered by Digifon itself. The list is available on the World Wide Web. Send e‐mail comments to immer@digifon.com. http://www.digifon.com ...
Page 312 EdNet San Francisco, CA 415‐274‐8800 Service to recording studios and broadcasters. Operates about five hundred sites, mostly at recording studios. They handle billing and booking, etc. Provides “translation” service. Offer full tech support, ISDN line ordering, etc. Full‐service Zephyr dealer. USER’S MANUAL Appendix 2 – Finding Public Codec User Lists...
Page 313: Appendix 3: Isdn Cause Phrases/Values
USER’S MANUAL Appendix 3: ISDN Cause Phrases/Values When there is an ISDN event or problem in the network, a cause code appears on the LCD screen or ISDN protocol log that communicates information about the cause of the trouble. See section 12.2 for how to enable ISDN logging through the serial or Telnet port. These “Cause Values” are numbers generated by the network, which the Xstream translates to the associated phrases. When possible, we use those suggested by the Bellcore standard. Note that in some cases there may be more than one meaning. This can frequently be evaluated by whether the message has been received by the calling party or the called party, or by the ISDN protocol being used. We have included the most common cause codes here. For a complete list, see http://www.telos‐systems.com/techtalk/default.htm . CLASS 1.1 (normal) Cause No. 1 ‐ Check number, redial (unallocated number) This cause indicates that the called party cannot be reached because, although the called party number is in a valid format, it is not currently allocated (assigned). Cause No. 2 ‐ No route to network /Prefix 0 dialed in error This cause indicates that the equipment sending this cause has received a request to route the call through a particular transit network which it does not recognize. The equipment sending this cause does not recognize the transit network either because the transit network does not exist or because that particular transit network, while it does exist, does not serve the equipment which is sending this cause. or This cause indicates the prefix 0 is invalid for the entered number Cause No. 3 ‐ No route to dest./Prefix 1 dialed in error This cause indicates that the called party cannot be reached because the network through which the call has been routed does not serve the destination desired. This cause is supported on a network‐dependent basis. or A 1 was dialed when not required. Redial without the 1. Cause No. 4 – No prefix 1/Special Information Tone The prefix 1 is not required for this number or ...
Page 314 USER’S MANUAL Cause No. 10 – Prefix 1 not dialed (DMS Custom only) Indicates that the prefix 1 was not included in the called party address and is required. Cause No. 11 – Excessive Digits Received, call is proceeding More digits were dialed than expected. Called number has been truncated to the expected number. Cause No. 14 – Excess digits received, call is proceeding More digits were dialed than expected. Called number has been truncated to the expected number. Cause No. 16 ‐ Far end disconnect (Normal call clearing) This cause indicates that the call is being cleared because one of the users involved in the call has requested that the call be cleared. Under normal situations, the source of this cause is not the network. Cause No. 17 ‐ Busy, try again later (User busy) This cause is used to indicate that the called party is unable to accept another call because the user busy condition has been encountered. This cause value may be generated by the called user or by the network. In the case of user determined user busy it is noted that the user equipment is compatible with the call. Cause No. 18 ‐ No far end response (No user responding) This cause is used when a called party does not respond to a call establishment message with either an alerting or connect indication within the prescribed period of time allocated (before timer T303 or T310 has expired). Cause No. 19 ‐ No answer (No answer from user, user alerted) This cause is used when the called user has provided an alerting indication, but not a connect indication within a prescribed period of time (before timer T301 has expired). Cause No. 21 ‐ Call rejected This cause indicates that the equipment sending this cause does not wish to accept this call, although it could have accepted the call because the equipment sending this cause is neither busy nor incompatible. This cause may also be generated by the network, indicating that the call was cleared due to a supplementary service constraint. Cause No. 22 ‐ Number changed This cause is returned to a calling party when the called party number indicated by the calling party is no longer assigned. The new called party number may optionally be included in the Appendix 3 – ISDN Cause Phrases/Values...
Page 315 USER’S MANUAL diagnostic field. If a network does not support this cause, cause no: 1, unallocated (unassigned) number shall be used. Cause No. 26 – Non‐selected user clearing This cause indicates the user has not been awarded the incoming call. Cause No. 27 – Destination out of order This cause indicates that the destination indicated by the user cannot be reached because the interface to the destination is not functioning correctly. The term "not functioning correctly" indicates that a signal message was unable to be delivered to the remote party; e.g. a physical layer or data link layer failure at the remote party, or user equipment off‐line. May also indicate local unit has incorrect SPIDs. Cause No. 28 ‐ Incorrect number (invalid number format, address incomplete)/Special intercept announcement This cause indicates that the called party cannot be reached because the called party number is not in a valid format or is not complete. or This cause indicates the user should be returned a Special Intercept Announcement. Cause No. 31 ‐ Network disconnect (Normal, unspecified)/Special intercept announcement: Call blocked because of group restrictions This cause is used to report a normal event only when no other cause in the normal class applies. or This value indicates that a user from outside a basic business group (i.e. Centrex) violated an access restriction feature CLASS 1.2 (Resource unavailable) Cause No. 34 ‐ No circuit available (circuit/channel congestion) This cause indicates that there is no appropriate circuit/channel presently available to handle the call. Cause No. 38 ‐ Net out of order This cause indicates that the network is not functioning correctly and that the condition is likely to last a relatively long period of time e.g. immediately re‐attempting the call is not likely to be successful. Appendix 3 – ISDN Cause Phrases/Values...
Page 316 USER’S MANUAL Cause No. 41 ‐ Net problem, redial (Temporary Failure) This cause indicates that the network is not functioning correctly and that the condition is not likely to last a long period of time; e.g. the user may wish to try another call attempt almost immediately. May also indicate a data link layer malfunction locally or at the remote network interface or that a call was cleared due to protocol error(s) at the remote network interface. Cause No. 42 ‐ Net busy, redial (Switching Equipment Congestion) This cause indicates that the switching equipment generating this cause is experiencing a period of high traffic. Cause No. 44 ‐ No channel available (Requested circuit/channel not available) This cause is returned when the circuit or channel indicated by the requesting entity cannot be provided by the other side of the interface. Cause No. 47 ‐ Resource unavailable/New Destination This cause is used to report a resource unavailable event only when no other cause in the resource unavailable class applies. or This cause is used to indicate that the original destination is unavailable and to invoke redirection to a new destination. CLASS 1.3 (service or option not available) Cause No. 50 – Requested facility not subscribed The cause is used to report that the user cannot use this feature because s/he has not subscribed to it. Cause No. 51 – Bearer capability incompatible with service request This cause indicates a user request for action was rejected because the action was incompatible with the capability of the call. Cause No. 52 – Outgoing calls barred This cause indicates that because of call screening provided by the network, the calling user is not permitted to make a call. Cause No. 54 – Incoming calls barred The user will not accept the call delivered in the SETUP message. Cause No. 57 – Bearer capability (Data/voice) not authorized. Appendix 3 – ISDN Cause Phrases/Values...
Page 317 USER’S MANUAL This cause indicates that the user has requested a bearer capability which is implemented by the equipment which generated this cause but the user is not authorized to use it. This is a common problem caused by wrong Telco provisioning of the line at the time of installation. Cause No. 58 – Bearer capability not presently available This caused indicates the user has requested a bearer capability which has been implemented by the equipment, but is not available at this time. CLASS 1.4 (Service or option not implemented) Cause No. 63 – Service or option not available, unspecified This cause is used to report a service or option not available, only when no other cause in this class applies. Cause No. 65 ‐ Bearer Capability not implemented (Incompatible bear cap) This cause indicates that the equipment sending this cause does not support the bearer capability requested. Cause No. 66 – Channel type not implemented This cause is returned when the called party has reached a channel type not supported. Cause No. 69 – Requested facility not implemented This cause indicates that the network (or node) does not support the requested bearer capability and therefore cannot be accessed at this time. Cause No. 79 – Service or option not implemented, unspecified This cause is used to report a service or option not implemented event only when no other cause in this class applies. CLASS 1.5 (Invalid message; e.g. parameter out of range) Cause No. 88 ‐ Incompatible destination This cause indicates that the equipment sending this cause has received a request to establish a call which has low layer compatibility, high layer compatibility or other compatibility attributes ...
Page 318 USER’S MANUAL Used to indicate that the destination address is missing or incomplete or Used to indicate a non‐existent Closed User Group Cause No. 95 – Invalid message, unspecified This cause indicates that the entity sending this cause has received and Invalid message, only when no other cause in this class applies. CLASS 1.6 (Protocol error; e.g. unknown message) Cause No. 96 – Mandatory Info missing This cause indicates that the equipment sending this cause has received a message which is missing an information element which must be present in the message before that message can be processed. Cause No. 97 ‐ Message type non‐existent or not implemented This cause indicates that the equipment sending this cause has received a message with a message type it does not recognize either because this is a message not defined, or defined and not implemented by the equipment sending this cause. Cause No. 98 – Message not compatible This cause indicates that the message received is not compatible with the call state or the message type is non‐existent or not implemented. Cause No. 99 – Info non‐existent or not implemented, call processed This cause is sent when the equipment sending this cause has received a message which includes the information elements not recognized because the information element identifier is not define or it is defined but not implemented by the equipment sending the cause. However, the information element is not required for the equipment sending the cause to process the message. Cause No. 100 – Invalid information element contents This cause is returned when the equipment sending this cause has received an information element that it has implemented; however one or more field of the information element are coded in such as way (e.g., truncated, invalid extension bit, invalid field values, etc) that has not been implemented by the equipment sending this cause. Cause No. 101 – Message not compatible with call state/Protocol error threshold This cause is returned when the equipment sending this cause has received a message that procedures indicate is not a permissible message to receive at this time. or ...
Page 319 USER’S MANUAL This cause indicates that the switch sending this cause is clearing the call because a threshold has been exceeded for multiple protocol errors during an active call. Cause No. 102 ‐ Timeout disconnect (Recovery on timer expiration) This cause indicates that a procedure has been initiated by the expiry of a timer in association with error handling procedures. Cause No. 111 – Protocol Error, unspecified This cause is used to report a protocol error event only when no other cause in this class applies. Among other things, this cause may be displayed if you failed to dial a “9” or “8” for an outside line, if required. Also may be returned if you have some types of restrictions as to the number of calls, etc. CLASS 1.7 (Interworking class) Cause No. 127 ‐ SW56 disconnect (Internetworking, unspecified) This cause indicates that an interworking call (usually a call to SW56 service) has ended. May also be seen in the case of a non specific rejection by your long distance carrier (try again at a different rate or with a different carrier) Appendix 3 – ISDN Cause Phrases/Values...
Page 320 USER’S MANUAL Appendix 3 – ISDN Cause Phrases/Values...
Page 321: Appendix 4: Known Working Spid Formats By Telco
Appendix 4: Known Working SPID Formats by Telco SPIDs are only used in the USA & Canada. Your SPIDs may be different! Note, for each line there is only one configuration that will work. Your SPID is distinct from your telephone number, and does not necessarily contain your area code or telephone number (although this is generally the case). A standardized SPID format for National ISDN of XXXYYYZZZZ0101 was phased in during 1996 and is used by most of the Telcos for new lines. Incorrect SPIDs will be indicated by a line status of “Wait” on the ISDN status screen. Certain Telcos will leave the last “00” or “01” off the end of your SPID when giving it to you. For the latest list check Telos Systems’ web page. Additional SPIDs can be found at INTEL’s SPID Page: http://support.intel.com/support/isdn/SPID_Tip.htm . XXX=Area code; YYY=NXX exchange; ZZZZ=phone number Alltel (Ohio) XXXYYYZZZZ0101 Ameritech XXXYYYZZZZ0111 XXXYYYZZZZ011 01YYYZZZZ011 XXXYYYZZZZ01 Bell Atlantic 01YYYZZZZ00 01YYYZZZZ000 01XXXYYYZZZZ000 XXXYYYZZZZ100 Bell Atlantic North (former NYNEX) Appendix 4 – Known Working SPID Formats by Telco USER’S MANUAL...
Page 322 Cincinnati Bell 01YYYZZZZ000 SPID1 00YYYZZZZ01 SPID2 00YYYZZZZ02 00YYYZZZZ01 Fort Mills Telephone XXXYYYZZZZ Frontier Communications 01YYYZZZZ000 GCI (Alaska) XXXYYYZZZZ00 XXXYYYZZZZ0101 XXXYYYZZZZ0001 01YYYZZZZ000 XXXYYYZZZZ00 01YYYZZZZ0 XXXXYYYZZZZ SPID1 XXXYYYZZZZ00 SPID2 XXXYYYZZZZ01 SPID1 XXXYYYZZZZ0000 SPID2 XXXYYYZZZZ0001 Appendix 4 – Known Working SPID Formats by Telco USER’S MANUAL SPID1 XXXYYYZZZZ0000 SPID2 XXXYYYZZZZ0101...
Page 323 SNET (Southern New England Telephone) 01YYYZZZZ00 01YYYZZZZ000 XXXYYYZZZZ0101 Southwestern Bell XXXYYYZZZZ0101 XXXYYYZZZZ 01YYYZZZZ000 01YYYZZZZ00 XXXYYYZZZZ01 Sprint/Centel XXXYYYZZZZ100 XXXYYYZZZZ0000 SPID1 XXXYYYZZZZ000 SPID2 XXXYYYZZZZ100 Standard Telephone (GA) SPID1 XXXYYYABCDCD SPID2 XXXYYYEFGHGH (last two digits of phone number repeated) Verizon- See Bell Atlantic, Bell Atlantic North, or GTE Verizon West- See GTE Vermont Telephone...
Page 324 USER’S MANUAL Appendix 4 – Known Working SPID Formats by Telco...
Page 325: Appendix 5: Suggested Reading And References
Appendix 5: Suggested Reading and References ISDN ISDN – A Practical Guide to Getting Up and Running; William A. Flanagan; Flatiron Publishing, Telecom Books, New York, 1996; www.telecombooks.com . Lots of nitty gritty deep technical information. ISDN for Dummies; David Angell; IDG Books, Foster City, CA; 1995. Good practical information. The ISDN Literacy Book; Gerald Hopkins; Addison Wesley, Reading PA, 1994. Lots of details on how ISDN came to be, how standards are written, etc. Telephony The Telephony Book – Understanding Telephone Systems & Services; Jane Laino; Telecom books, a division of Miller Freeman, USA, 1999; www.telecombooks.com . This book is a general introduction however it is a good starting point for those new to Telephony. ABC of The Telephone –Anatomy of Telecommunications; ABC Teletraining; www.abcteletraining.com . A great introduction to all aspects of Telephony. National Association of Broadcasters Engineering Handbook (10th Edition); National Association of Broadcasters, Washington, 2005; www.nab.org . Chapter 3.10 is written by our founder, Steve Church and contain a lot of new material. It will be available on the Tech Talk section of our website at: http://telos‐systems.com/ Newton’s Telecom Dictionary (21 Edition and counting) Harry Newton; Telecom books, New York; www.telecombooks.com . This book is more of an Encyclopedia than a dictionary. It is well worth the space on your bookshelf. Understanding Telephone Electronics, Fourth Edition; Stephen J. Bigelow, Joseph J. Carr, and Steve Winder; Newnes, Boston, 2001. A somewhat more technical introduction to telephony. Focuses on POTS lines and terminals, but also covers digital theory and transmission. Wiring (Network and Telephone) Mike’s Basic Guide to Cabling Computer’s and Telephones; Mike Gorman; Prairie Wind Communications, Inc, 1997. This book is destined to be a classic. Its down to earth practical style, copious illustrations, and practical information covers all bases. Technician’s Handbook to Communications Wiring; Jim Abruzzino; CNC Press, Chantilly VT, 1999. This small book is concise, yet contains a lot of great information including proper technique for working with Cat. 5 cable and connectors. The Axia Guide to Choosing Category Cable; Stephen Lampen, Belden Electronics. Available for download at: http://www.axiaaudio.com/tech/cable/default.htm ...
Page 326 USER’S MANUAL IP and Livewire Introduction to Livewire – an Axia Audio publication. Available for download at: http://www.axiaaudio.com/manuals/files/IntroToLivewire2.1.pdf Practical IP and Telecom for Broadcast Engineering and Operations; Fred Huffman A title in the Focal Press Media Technology Professional Series (2004) http://www.elsevierdirect.com/product.jsp?isbn=9780240805894 Other Murphy’s Law and Other Reasons Why Things Go Wrong!; Arthur Block; Price/Stern/Sloan Los Angeles; 1977. Essential for keeping proper perspective while troubleshooting. The Dilbert Future‐Thriving on Stupidity in the 21 Century; Scott Adams; Harper Business. New York; 1997. This book provides interesting perspective on ISDN. Appendix 5 – Suggested Reading...
Page 327: Appendix 6: Streaming And Special Operational Modes
USER’S MANUAL Appendix 6: Streaming and Special Operational Modes The Zephyr Xstream offers several special modes that, when selected, change its functionality, as well as its "personality". Ethernet (IP) modes allows transmission and reception of audio streams over IP networks. Ideal for casual monitoring, or audio distribution over specially managed IP networks. Operation over xDSL or other IP based networks is possible in this mode. Error concealment is provided in AAC mode. V.35 Single mode allows high‐bitrate operation over synchronous data transport mechanisms including T1 (E1) and fractional T1 (E1). We have deliberately separated these two modes as the correct settings for these applications (Internet streaming versus real time audio transmission over ISDN versus high‐speed synchronous data channels) are different, and we thought that it would add unnecessary confusion to list all options when a number would not be useful for a given situation. ETHERNET MODES The Zephyr Xstream's Ethernet mode is used to generate streaming MP3 or AAC feeds. The Xstream supports Unicast operation only, however depending on the bitrate, multiple streams are permitted. This mode is selected by changing the Interface selection in the third TEL menu to Ethernet(SIP), or Ethernet (RTP). Making this change will change the options for the CODEC menu. The Dial screen and first TEL screen will also change, showing status of the Streaming IP connections. THE CODEC MENU IN ETHERNET MODE The Codec menus will be different from those shown elsewhere in this manual when the Interface selection (in the 3rd TEL menu) has been set to Ethernet. CODEC- Screen 1 (Status) This screen shows the encoding algorithm, bit rate, mode (mono or stereo) and sample rate currently being used for encoding.
Page 328 Transmit The Transmit setting determines what standard (i.e. Algorithm) is used to encode the audio to be sent to the far end. Each mode offers distinct advantages and disadvantages (see section 6.2 for additional information), as follows: AAC – THE MPEG AAC algorithm will be used L3 – THE MPEG Layer 3 algorithm will be used Bitrate This allows selection of the desired bit rate for the streaming audio. The higher the bitrate, the better the fidelity. However, higher bitrates use up more bandwidth thereby increasing your transmission cost. See below. On the other hand, the QOS (quality of service) of the IP network is less critical the lower the bit rate used. Disconnect all clients This allows you to terminate the connections to Xstreams (or other players) currently listening to the stream generated by the local Xstream THE TEL MENU IN ETHERNET MODE TEL - Screen 1 (Status) This screen shows the number of client players (software players or Zephyr Xstreams) currently connected to the stream generated by this Xstream. This information is valuable as dropouts are more likely the greater the number of connected clients The remaining TEL screens are identical in Ethernet mode as in ISDN or V.35 mode. INTERNET BANDWIDTH REQUIREMENTS The Xstream can serve a limited number of streams without a replication server. The following table indicates the allowable number of streams: USER’S MANUAL Appendix 6 – Special Operational Modes...
Page 329 Bit Rate (per stream) Allowable Streams The formula for estimating the required bandwidth requirement is: Required Bandwidth = Number of Unicast Listeners x (encoded audio stream bitrate x 1.1) The following table shows estimated numbers of listeners for each Internet connection style, assuming a 16Kbps encoded audio bitstream rate (this does not imply this number streams are supported by the Xstream, however): Connection Type ISDN Basic Rate Ethernet LAN (10Mbit) Ethernet (100Mbit) Of course, you may not wish to allocate all of your available bandwidth to audio streaming. STREAMING DEFINITIONS Pull Only The connection in this type of stream is initiated by the listener (client). The client contacts the sender (server), requesting a stream to be sent. The server sends a stream to the client, and the client sends no further messages to the server until the user disconnects. At disconnect time, the client sends a message to the server requesting that no more data be sent. Push Only The audio stream, in this instance, is initiated by the sender. The user specifies the client to which to send the stream, and the data is sent without any negotiation. The client decodes the audio it is sent, but has no control over when it starts or stops. The stream must be disconnected by the send‐side user. Bi‐directional Uses some combination of the above to initiate a stream in both directions. The user at either end can initiate the connection, and the software will negotiate both the push and the pull involved. Either end's user can likewise disconnect the call. USER’S MANUAL Bandwidth Listeners...
Page 330 STREAMING MODES Depending on the Interface selected in the TEL menu, the Xstreams streaming works differently. Here is a review of the different options: HTTP The familiar Pull‐Only HTTP remains, however we strongly recommend the use of one of the following RTP based options. The Zephyr is still a server only, not a client. HTTP streaming is used primarily with desktop players (Audioactive Player, WinAmp, etc.) for confidence audio. It is not to be used on the public internet due to limitations in the Zephyr's TCP stack. If long‐range/long‐term HTTP streaming is required, a Shoutcast Server (or other stream replicator) can be used as a client. RTP/UDP Real‐Time Protocol streams are Push‐only. One server can send to several clients, who decode the audio if they are also in RTP streaming mode. RTP is a transport protocol, not a connection protocol, which is why it works in this way. The "connection protocol" is the user hitting the Dial button. SIP Unlike RTP, SIP is a connection protocol, not a transport protocol. SIP negotiates a bi‐directional connection between the two units, behaving much like a telephone or ISDN call. Progress is related to the far side as well as the local user. The far end unit may be reported as busy (unlike RTP, which sends regardless of the readiness of the receiver), etc. SIP uses RTP as its transport protocol. What this means, basically, is that SIP negotiates for each side of the connection to send an RTP, push‐only stream to the other side. The result is a bi‐directional connection. PORTS (IP PORT) Each IP device must have an IP address. However, each IP address can handle multiple connections, using different ports. Sort of like different extensions on the same office telephone system. IP ports to be used by the streaming system are listed on the Telco configuration page in all modes. USER’S MANUAL Appendix 6 – Special Operational Modes ...
Page 331 SIP Port (Default 5060) SIP uses a TCP port, as its communication is bidirectional. The SIP port is what is used for the connection protocol, that layer that initiates the two RTP streams. The SIP port must match on both ends of a connection. UDP Port (Default 9150) RTP streams are sent and received over the UDP port. This port is used for both RTP and SIP modes, as SIP negotiates the creation of RTP streams. The UDP port must match on both ends of a connection. HTTP Port (Default 8080) The TCP port used for HTTP streaming only (normal web‐pages are also served over HTTP, but these are sent over port 80, the usual port to which a web browser connects). A player which supports HTTP streams can connect using the usual port notation (e.g. http://www.xxx.yyy.zzz:PORT). Note that prior to 3.0.1p, the Zephyr Xstream used port 8000 for HTTP streaming. TCP Port (Default 8800) This port is currently unused. FIREWALLS AND ROUTERS The Internet works because there is a central authority for doling out IP addresses. Any given connection to the internet is permitted to use some number of designated addresses which can be accessed by any machine that is also connected to the Internet. Most connections are limited to one IP, with many more machines needing access to the network. This is the basic purpose most routers. When a device, such as the Xstream has a "public" (e.g. real) IP address it can be directly accessed through a router. However, if there is a firewall (which may be built into the router, or a separate box), or a "private" IP address the unit is not directly accessible. In these cases there are a few options, as outlined below. Port Forwarding Port forwarding is very simple. When you have two connected networks, with different numbering schemes, they must be connected by a router. The machines on either side of the router cannot access each other. The router itself can be accessed by machines on either network, however. USER’S MANUAL Appendix 6 – Special Operational Modes...
Page 332 Port forwarding is basically telling the router that when it receives packets from the Wide Area Network on a given port, from those packets should be forwarded transparently to the specified address on the Local Area Network. NAT (Network Address Translation) This is a scheme by which a private network can make connections into a public network (such as the Internet) using only one address on the public net. An example is having 3 computers behind a router/firewall on a residential DSL line. The three computers have separate addresses on the local network, but only the Router's WAN IP is visible on the outer network. NAT works because the router knows the format of TCP/IP packets. It can substitute its own IP into the packets that are outbound, then reinsert the proper LAN IP when a packet comes inbound from the WAN. This works because TCP is connection‐based, so the router knows that all traffic on a given connection is destined for the machine that initiated it. THE ZEPHYR XSTREAM BEHIND A FIREWALL In our world, we will always have a studio codec and a remote codec so there are always two Xstreams to consider. For this discussion, we'll assume that the remote Zephyr Xstream is properly configured already. That way, we only need to discuss the considerations for the local machine. If each operator does this, both codecs will end up being configured properly. To begin, for all of the scenarios presented below, the following procedure should be followed: • Give the Zephyr a fixed IP address on the LAN. Usually, this is done by simply picking an unused address. • Set the LAN subnet mask (usually 255.255.255.0 ‐ get these numbers from the IT administrator) • Set the gateway to the LAN address of the router • Set the DNS to the server specified in the WAN/DHCP area of the router's information pages (optional). HTTP Mode Not recommended for use over the public network. If it's behind a router, it really shouldn't be used. ...
Page 333 Contact the operator on some other channel to confirm receipt of the stream, as there is no reverse communication Receiving To receive an RTP stream from behind a firewall, you first need to set that firewall for port forwarding. On the router setting page, forward UDP port 9150 to the IP address the receiving Xstream. Contact the remote operator, and provide the WAN address of the router (retrieved from the router status page or IT admin). When the remote operator sends a stream to your machine, the router will automatically forward it to your Xstream. Note that you can only forward a port to one IP at a time, so if you are behind a firewall and you need to receive multiple streams, you'll have to forward other ports and have the remote admin change their UDP port to match yours. SIP Mode Since SIP mode is bidirectional, we'll discuss Dialing and Answering, rather than Sending and Receiving. Answering After configuring your Xstream, you'll need forward TCP port 5060 to the Xstream you wish to use. When this is done, follow the instructions for RTP Mode (Receiving) ‐ forward UDP 9150 to the Xstream and give the remote operator the router's WAN address. Dialing Configure your Xstream as with SIP Mode (Answering). With your unit in Ethernet (SIP) mode, go to Codec(3) – Ethernet Settings. The field "RouterIP" (Renamed in current development build as "WAN IP") should be set to the router's WAN address, as shown on the router's status page. SIP includes IP information in its own protocol, above the TCP/IP layer, which cannot be automatically changed by the NAT router. If there is no router, or the Xstream is on a publicly‐available IP address, the WAN IP should be set to 0.0.0.0. USER’S MANUAL Appendix 6 – Special Operational Modes...
Page 334 TRANSMITTING AN MPEG Layer 3 or AAC STREAM Changing the Interface selection in the 3rd TEL menu to Ethernet will automatically start a stream. The bit rate and Transmit (coding mode) for that stream are determined in the CODEC menu (see above). The party(ies) connecting to this stream will need your IP address to connect to it. RECEIVING IN ETHERNET MODE – The DIAL Menu Connecting to a Stream from another Xstream 1. Press <DIAL> 2. Enter the IP address of the desired stream (use the <*> or <#> key of the "." character. 3. Press <DIAL> again IMPORTANT TIP! The system will automatically determine the coding mode and bit rate automatically. HOT TIP! IP addresses do not include leading Zeros If you are connecting to an IP address given to you as 012.345.000.789 the correct IP address would be 12.345.0.789...
Page 335 V.35 SINGLE Mode (high speed V.35/X.21 synchronous operation) The menus and operation of the Xstream when using the V.35 Split setting of the Interface selection in the third TEL menu are identical to those for ISDN option for that parameter. This is documented in the main body of this manual. When V.35 Single mode is set for Interface selection in the third TEL menu, certain menus change. We will document those settings here. The V.35 Single mode is appropriate for situations where a single synchronous network connection at 96, 112, 128, 256, or 368kbps is available via a V.35 or X.21 interface. HOT TIP! Layer 3 offers operation at 112, 128, and 256 kbps only. This is because at 384 kbps Layer 2 or AAC is preferred over Layer 3. 96 kbps operation is offered only in AAC.
Page 336 This encodes a single mono bitstream using a single audio input (the audio on channel A). L3 mono offers 15 kHz audio frequency response if the Sample Rate is set to 32 or 20 kHz audio frequency response if the Sample Rate is set to 48 kHz. HOT TIP! If you wish to feed a mix of the audio on channels A and B, you can do so by setting up the send mixer section (MX/MXP models) appropriately, or by setting the Send Mix option to "A + B"...
Page 337 These work similarly to the equivalent Layer‐3 modes. Offers 20 kHz audio frequency response when the sample rate is 48 kHz and 15 kHz response when the sample rate is set to 32 kHz. L2 Mono 128 This mode is very common mode used in the voice‐over industry. It works in the same way as the other Mono 128 modes. Offers 19 kHz audio frequency response when the sample rate is set to 48 kHz and 15 kHz response when the sample rate set to 32 kHz. Receive: The appropriate Receive mode will be determined by the encoding or Transmit mode used at the far end. The Xstream's Receive modes in V.35 Single mode are: L3 Receives all Layer‐3 modes available in V.35 Single mode. L2 Receives all Layer‐2 modes available in V.35 Single mode. AAC‐LD Receives all AAC‐LD modes available in V.35 Single mode. AAC Receives all AAC modes available in V.35 Single mode. BIT RATE This determines the bit rate of the V.35 ports as well as the encoder and decoder. In V.35 Single mode the following rates are available: 96, 112, 128, 256, 384 kbps. Not all Bit Rates are supported in all algorithms. SAMPLE RATE This determines the sample rate used for the codec operation. Not all sample rates are supported at all bit rates. The following table summarizes the allowable sample rate and bit rate combinations supported in Single V.35 mode. USER’S MANUAL Appendix 6 –...
Page 338 ALGORITHM AAC Stereo 32 kHz 48 kHz AAC Mono 32 kHz 48 kHz AAC-LD Stereo 32 kHz 48 kHz AAC-LD Mono 32 kHz 48 kHz Layer 2 Stereo Supported Layer 2 Mono 32 kHz 48 kHz Layer 3 Stereo 32 kHz 48 kHz Layer 3 Mono 32 kHz...
Page 339 USER’S MANUAL Appendix 7: Ordering ISDN for ETS 300 (Euro ISDN) This section applies to users in Europe and other countries where the Euro‐ISDN ETS 300 standard is supported (Europe, much of South America, Australia, South Africa, Mexico, etc). Since there are virtually no options when ordering Euro ISDN, this section is probably not needed, however we thought we could help by summarizing what you will require. If you live in a country outside of North America that does not support Euro‐ISDN, your local Telos Systems representative should be able to provide you with information about ordering (and using) ISDN in your area. We assume you have read the description of ISDN in Section 7 (ISDN Basics) and are familiar with ISDN’s basic concepts. If you have not read that section, doing so now will help you to better understand the information that follows. On one ISDN BRI circuit, there are three simultaneous bi‐directional channels: two 64kbps “bearer” or “B” channels for the transmission of user information and one 16kbps “data” or “D” channel for call set‐up and status communication. That is why this is sometimes called 2B+D service. You can use the B channels, but only the phone companies and equipment manufacturers can usually use the D channel. From the perspective of telephone network routing, each channel appears to be a separate line with its own number and independent dial‐out capabilities. Since each has to be dialed or answered separately, they appear to be “lines” to users also. We refer to a B channel as a “line” on the Zephyr Xstream’s menu. ISDN comes into your facility on two wires, just like plain old telephone service (POTS). There the Telecom will install a NT1 (Network Termination) to provide you access to the S (4‐wire) interface usually provided. You will be connecting this to the Zephyr Xstream’s top (8‐pin) jack. Details, Details In order to communicate accurately what it is you need, we think you should be familiar with the vocabulary used to describe ISDN. As with anything, for best results, it helps to know what you are talking about. For an in‐depth glossary of terminology please check Appendix 9 or the Telos web site at http://www.telos‐systems.com . Names We’ve seen Basic Rate Interface ISDN called by a number of names. The term “S∅” is ...
Page 340: Appendix 7 - Ordering Isdn For Countries Using Ets300 (Euro Isdn)
USER’S MANUAL CSD and CSV Recall that each ISDN BRI has two possible B channels. It is possible to order a line with one or both of the B channels enabled – and each may be used for voice and/or data use. Phone terminology for the class of service is CSV for Circuit Switched Voice connections and CSD for Circuit Switched Data connections. (In contrast to PSD, Packet Switched Data, this is possible, but irrelevant to our needs.) CSD is used for codec connections. Even though you may be sending voice, the codec’s output will be in data form and therefore must be handled transparently by the telephone network. This happens because the Zephyr Xstream indicates that this is a CSD call (over the D channel) when the call is set up. CSV is for interworking with voice telephone service, and allows ISDN to interwork with analog phone lines and phones at the far end. CSV is required when communicating to a Zephyr Xport that is on a POTS line. You will be ordering one or more ISDN BRI 2B+D circuits with CSD (required) and CSV on each B channel. Euro ISDN normally provides both CSV and CSD. NT1s The ISDN standard specifies two reference points, the “U” and the “S” interfaces. The U is the single‐pair (2‐wire) bare copper from the Telco CO. A device called a “Network Termination, Type 1” converts this to the two‐pair (4‐wire) S interface. The Zephyr Xstream includes a built‐in NT1. This is for use in the USA & Canada. You should not connect to the U interface in other countries without first consulting the Telco. They may have regulations against doing so, and they will probably provide power for remote powering an NT1, and this power could damage your Zephyr Xstream’s ISDN interface. The older Zephyr classic units sold outside the USA may only have an S interface. If you plan on taking those units to Canada or North America you will need to rent or purchase an external NT1 for use while in those countries. One rental source is Silver Lake. See their website at www.silverlakeaudio.com . PP vs PMP The S interface allows for multiple terminals to be connected simultaneously. This is called Point to Multi‐Point (PMP) operation. When a single terminal is used this is referred to as a Point to Point configuration (PP). The Xstream can be used in either environment. If you intend to have other terminals share the S interface with the Xstream, you should make sure your Telco configures their switch properly to support this (PMP). You will also need to follow the correct termination rules if you will have multiple terminals. ...
Page 341 until a call attempt is made. If your application is critical, you may wish to request that the Telecom disable line deactivation. FA XAB L E ISDN ORDER FORMS FOR ETS 300 Following form may be used to place orders for ISDN lines in countries using the ETS 300 protocol. These should give the Telecom all of the information they need. Complete the top portion of the form and send it to the Telco. Appendix 7 – Ordering ISDN for Countries Using ETS300 (EURO ISDN) USER’S MANUAL...
Page 342 Long distance carrier: Date needed: We request an ISDN Basic Rate Interface (ISDN-2, BRI, S∅) circuit for use with the Telos Zephyr Xstream digital audio transmission system. This device interfaces audio equipment to digital telephone services. It requires Circuit Switched Data (CSD). We may also use this to makes calls using Circuit Switched Voice (CSV) service.
Page 343: Appendix 8: Ordering Isdn For The Usa & Canada
USER’S MANUAL Appendix 8: Ordering ISDN for the USA & Canada This section applies to users in Canada & USA and is intended to help you when it comes time to order your ISDN lines. We assume you have read the description of ISDN in section 7 (ISDN Basics) and are familiar with ISDN’s basic concepts. If you have not read that section, doing so now will help you to better understand the information that follows. Because ISDN in the USA is complex, there is occasionally confusion about what it is and how to get it. If you know who to call and provide them with complete information, the ISDN ordering process can be simple and straightforward. This guide takes you through the process step‐by‐ step. Telos Technical Support is available to answer any further questions you may have. Don’t take “no” for an answer if your Telco does not understand your needs. On one ISDN BRI circuit, there are three simultaneous bi‐directional channels: two 64kbps “bearer” or “B” channels for the transmission of user information and one 16kbps “data” or “D” channel for call set‐up and status communication. That is why this is sometimes called 2B+D service. You can use the B channels, but only the phone companies and equipment manufacturers can usually use the D channel. ISDN comes into your facility on two wires, just like plain old telephone service (POTS). From the perspective of telephone network routing, each channel appears to be a separate line with its own number and independent dial‐out capabilities. Since each has to be dialed or answered separately, they appear to be “lines” to users also. We refer to a B channel as a “line” on the Zephyr Xstream’s menu. The Zephyr Xstream’s internal ISDN interface (sometimes called by its generic name “Terminal Adapter”) is used to connect to digital ISDN telephone circuits. It easily adapts to the various types of service offered by the range of Central Office (CO) switches installed by telephone companies in the USA and Canada. Ordering ISDN Dealing with the Phone Company As is often the case when broadcasters interface with phone people, the lines of communication ...
Page 344 COMPANY TELEPHONE NUMBER Ameritech 800‐TEAMDATA (800‐832‐6328) Bell Atlantic 800‐570‐ISDN (800‐570‐4736) (see below for former NYNEX regions) Residential 800‐204‐7332 Bell Atlantic North Call your account representative. If you do not know who s/he is, call; (former NYNEX ) 800‐GET‐ISDN (800‐438‐4736) Bell South 615‐401‐4347 (Lorilynn Smith) Cincinnati Bell 513‐566‐DATA (513‐566‐3282) GTE 800‐GTE‐4WCN (800‐483‐4926) Natco 800‐775‐6682 ext 288 Nevada Bell Small Business 702‐333‐4811 ...
Page 345 USER’S MANUAL Details, Details In order to communicate accurately what it is you need, we think you should be familiar with the vocabulary used to describe ISDN. As with anything, for best results, it helps to know what you are talking about. For an in‐depth glossary of terminology please check Appendix 10 (Glossary) or Telos’ web site at http://www.telos‐systems.com . Protocols In a perfect world, all ISDN terminal equipment would work with all ISDN lines, without regard for such arcana as 5ESS, DMS100, CSV/CSD, SPIDs, etc. The USA ISDN “standard” had been evolving for several years. The good news is that it has now settled down, with most new installations using National ISDN‐1 or National ISDN‐2 (see below). At their central offices, the telephone companies use AT&T/LUCENT 5ESS, Northern Telecom DMS100, or Siemens EWSD switching equipment. While each will work with the Zephyr Xstream, there are some differences, which need to be taken into account when lines are ordered and used. Each has a “protocol” – the language the user equipment and the telephone network use to converse (on the D channel) for setting up calls and the like. There are standard protocols that all switches may provide, called National ISDN‐1 (NI‐1) and National ISDN‐2 (NI‐2). These protocols were standardized and specified by Bellcore, the technical lab jointly owned by the phone companies. However, both AT&T and Northern Telecom had versions of ISDN which predated the NI‐1 standard and a few switches may not been upgraded to the new format. In that case you would need to use one of the older "custom" protocols. There is also a newer NI‐2 standard, but it is designed to be compatible with NI‐1 for all of the basic functions and therefore is acceptable to device such as the Zephyr that use NI‐1. SPIDs Service Profile Identification (SPID) numbers are required in all but one of the AT&T protocols. These numbers are given to the user by the phone company and must be entered correctly into the Zephyr Xstream in order for the connection to function. SPIDs usually consist of the phone number plus a few prefix or suffix digits (it may or may not include the area code). There is frequent confusion between phone numbers and SPIDs, even among Telco personnel. While the SPID frequently includes the corresponding phone number, this is not necessarily the case. The intention of the SPID is to allow the Telco equipment to automatically adapt to various user requirements, by sensing different SPIDs from each type or configuration of user terminal. None of this matters with our application, but we must enter the SPIDs nevertheless. BellSouth has proposed, and most of the other telephone companies have agreed to implement, a standardized SPID for new installations. This has been implemented by most Telcos. The standard is area code+phone number+0101 (XXXYYYZZZZ0101). See Appendix 4 for a list of SPID formats used by the Telcos prior to the standard. ...
Page 346 USER’S MANUAL providing you with the phone numbers of your B channels (often called "Directory Numbers", or "DNs") and your SPIDs! Directory Number (DN) These are simply your seven digit phone numbers, as would be listed in your local telephone directory, and need to be entered into your Zephyr Xstream. Normally each ISDN BRI circuit will have two DNs and two SPIDs. Each DN corresponds to a given SPID and when you enter this information, you must enter them to reflect this relationship. CSD and CSV Recall that each ISDN BRI has two possible B channels. It is possible to order a line with one or both of the B channels enabled – and each may be enabled for voice and/or data use. Phone terminology for the class of service is CSV for Circuit Switched Voice and CSD for Circuit Switched Data. (In contrast to PSD, Packet Switched Data, which is possible but irrelevant to our needs.) CSV is not required for Telos Zephyr Xstream hi‐fidelity audio connections. Even though you may be sending voice, the codec’s output will be in a data form and must be handled transparently by the telephone network. This happens because the Zephyr Xstream indicates that this is a CSD call (over the D channel) when the call is set up. CSV is for interworking with voice phone service and allows ISDN to interwork with analog phone lines and phones. CSV is required to place calls to a Zephyr Xport that is using a POTS line. You will be ordering one or more ISDN BRI 2B+D circuits with CSD (required) and CSV (optional) on each B channel. Alternate CSD/CSV should be requested. NT1s The ISDN standard specifies two reference points, the “U” (2‐wire) and the “S” (4‐wire) interfaces. The U is the single‐pair bare copper from the Telco CO. A device called a “Network Termination, Type 1” converts this to the two‐pair S interface. In Europe, Asia, South America, and most of the world, the NT1 is always provided by the phone company. The Zephyr Xstream provides both the U interface (on an 6‐position 4‐pin miniature “RJ‐11 style” modular connector) and the S interface (on an 8‐position 8‐pin miniature “RJ‐45 style” modular connector). Terminals and Terminal Types Any equipment connected to an ISDN line is a ‘terminal’ – whether phone, computer, or codec. ...
Page 347 FA XAB L E ISDN ORDER FORMS For the USA & Canada Following forms should be used to place orders for ISDN lines in the USA & Canada. Since ISDN is very complex in those countries, it is necessary to use these forms or you may experience problems. These should give the phone company all of the information they need. The majority of installations, if ordered in writing, with this information, go smoothly. If you do experience problems Telos technical support is here to help. You may also wish to look at our troubleshooting information in the Zephyr Xstream manual. Complete the top portion of the form and send all three pages to the phone company. Keep the form handy and show it to the installer when he or she puts in your line and ask the installer to verify with the switch programmer (at the central office) that your ...
Page 348 Long distance carrier: Date needed: We request an ISDN Basic Rate Interface (BRI) circuit for use with the Telos Zephyr Xstream digital audio transmission system. This device interfaces audio equipment to digital telephone services. It requires Circuit Switched Data (CSD). We may also use this device to make calls using Circuit Switched Voice (CSV) service.
Page 349 PROTOCOL: National ISDN-1 (or National ISDN-2) From AT&T 5ESS, Northern Telecom DMS100, and Siemens EWSD. If you use IOC Capability Packages, please use Capability Package ‘S’ unless otherwise specified. CO Values Line Type: National ISDN-1 Bearer Service: CSD/CSV on both channels (CSD is required; CSV may be omitted if requested by user) TEI: One dynamic per number Terminal Type: A 101XXXX Long distance access: Yes...
Page 350 ASDFI267 (Prohibit REX on BRI lines per line) Give your customer the following information: 1) One Directory Number PROTOCOL: AT&T Point-to-Multipoint (Custom) Not supported by the Telos Zephyr Xstream. Please provide National ISDN or AT&T Point-to-Point (Custom) Appendix 8 – Ordering ISDN for the USA and Canada...
Page 351: Appendix 9: Modular Cable Guide
Appendix 9: Modular Cable Guide This appendix is intended to be a reference tool that covers various modular telephone‐ style connectors used with your Xstream. IMPORTANT! Be very careful to label wiring using RJ- style connectors. In some cases, they have power on them, and there is a risk of applying this power to something that does not expect it.
Page 352 Note that TIA/EIA‐ 568‐ A T568B is electrically equivalent, however the Green and Orange pairs are reversed as follows: TIA/EIA‐568‐A T568B color codes. Note that this is electrically Identical to TIA/EIA‐568‐A T568A, however the green and orange pairs are switched. Hot Tip! While TIA/EIA- 568- A T568A and TIA/EIA- 568- A T568B are electrically equivalent, care must be taken that both ends of a given cable utilize the same system. For that reason, we strongly recommend choosing one of the two standards and using It throughout your facility.
Page 353 This special Ethernet "crossover" cable is used when two computers (or a computer and another device such as a Zephyr Xstream) need to be connected without using an Ethernet hub. ISDN U interface Cable Pin‐out The U interface, specified in the USA & Canada, is specified between the Telco and the user‐ provided NT1. The U interface is a robust interface meant for use on existing Telco copper plant. It is a single copper pair. It is provided on the center pair of either a 6‐ position 4‐ pin miniature modular jacks (RJ‐ 11 style) or a 8‐ position 8‐ pin miniature jack (RJ‐ 45 style). The standards specify the compatibility between the 6‐ position plug and 8‐ position jack so either can be used. For this reason, we prefer the 6‐ position connectors on cables, as they will work with either type of receptacle. ISDN U interface cable (typical) Typical ISDN U interface cable. This is identical to a POTS telephone cord. Polarity should not matter, so a straight through cable is also acceptable. ISDN U interface connector (USA & Canada only) ISDN U interface connector. Note, the polarity of the line connections should not matter. USER’S MANUAL FUNCTION Not Present No Connection Line (R) Line (T) No Connection Not Present FUNCTION No Connection No Connection Line (R) Line (T) No Connection...
Page 354 Blue White/Blue Orange White/Brown Brown Note A – Not used in Telos products Note B – Optional, not required if remote PS2 power is not needed ISDN S interface cable wiring diagram. Both ends are wired identically. HOT TIP This cable has 4 twisted pairs wired “straight through” just like many cables normally used for Ethernet 10Base- T.
Page 355 USER’S MANUAL FUNCTION No Connection No Connection S Transmit to network + S Receive from network + S Receive from network - S Transmit to network - PS2 Power (- 48 VDC) PS2 Power ground (+) The S interface is a standard modular RJ‐45 style jack with the pin‐out shown above. Appendix 9 – Modular Cable Guide...
Page 356 USER’S MANUAL Appendix 9 – Modular Cable Guide...
Page 357: Appendix 10: Glossary
USER’S MANUAL Appendix 10: Glossary You’ll get better results from the Telco if you understand, and speak, the lingo! We have tried to include the typical acronyms used by Telco personnel. We’ve put the definition under the most commonly used acronym. The most up‐ to‐ date version of this glossary is available at the Telos web site www.telos‐ systems.com AMI Alternate Mark Inversion. A T1 line coding method. This is the older of the two commonly available. See line coding, T1. See Also B8ZS. ANI – Automatic Number Identification A system, originally designed for use by Interexchange carriers (IECs), that transmits the “billed party number” along with a call. Note that the billed party number is not necessarily the number of the line placing the call. ANI predates SS7 and can operate with analog as well as digital trunks. See also CLID and Caller ID. Asynchronous Data A form of serial data communication that is not clocked. To keep the bit stream synchronized, start and stop bits are added, which cuts down on throughput. RS‐ 232 computer data is commonly asynchronous data. In contrast to synchronous data. B Channel ‐ Bearer Channel. One of the multiple user channels on an ISDN circuit. Used to carry user’s data; i.e. coded audio data in the case of Zephyr or ZephyrExpress. B8ZS‐ Bipolar 8 (with) Zero Substitution. A T1 line coding method. This is the more modern line coding method of the two commonly available. See Line Coding, T1. See also AMI Bearer Channel See B Channel Behind the PBX This is our own term, and refers to when one privately owned phone system is tied to another privately owned phone system. The most common application is when a key system is connected to analog ports of a PBX. When it involves one PBX behind another, it is a limited Tandem application. See Tandem Switch and Tandem Tie Trunk Switching below. Bell Labs. The basic research facility that was AT&T’s primary research facility. Bell Labs was originally part of Western Electric and was spun off with Lucent Technologies when AT&T was broken up in 1984. Many very important discoveries were made at Bell Labs including the transistor, communications theory, and radio astronomy. The future of Bell Labs seems bleak at the time of this writing. Bell Operating Company. See BOC. See also RBOC. Bellcore‐ BELL COmmunications REsearch. See Telcordia. The research and development organization owned by the RBOCs. Bellcore represents the RBOCs in developing standards for Telco equipment and in testing equipment compliance to those standards. Bellcore also offers educational and training programs open to all interested parties. Now Telcordia. See Telcordia BERT ‐ 1) Bit Error Rate Test A test for digital lines which involves looping a data path and sending a test pattern. Data returning is compared to the sent data to check for errors. Depending on the “Test Pattern” used, BERTs may or may not uncover problems. A line, which ...
Page 358 USER’S MANUAL Billing Telephone Number The main phone number which all calls on hunt group or a PRI are billed to. This information may be required when configuring a PRI PBX. Bit Error Rate The basic measure of errors on digital transmission paths. It is usually expressed as the number of errors per number of bits. For example, the allowable bit error rate on a BRI circuit is 1 x 10 E7 (one bit error in 107 bits or 1 error in 10 million bits). Bit Error Rate Test See BERT Bit Rate The capacity of a digital channel. ISDN calls are set up at a given bit rate, either 64Kbps or 56Kbps. The bit rate of a circuit‐ switched connection cannot be changed during a call. See Kbps. BLEC Building Local Exchange Carrier. A LEC who covers the occupants of a single building (or a small group of buildings) only. Often Telecom services are provided by a BLEC as a service or incentive to potential tenants. If a BLEC offers Long Distance Service it is covered by the same regulations as any other LEC. Blocking When a circuit switched call cannot be completed. The percentage of blocked calls to the number of calls attempted forms the basis of a statistic called “grade of service”. While it is economically infeasible to build a network that would have no blocking, the Telcos are held accountable by the utility commissions to keep blocking below tariffed levels. The concept of blocking cannot be applied to packet networks (they just lose packets, instead), only circuit switched networks. Blue Alarm Also called an Alarm Indicating Signal (AIS). A keep‐ alive signal sent if a problem occurs mid‐ span in a T‐ carrier system. The blue alarm signal is required because in some cases T‐ 1 repeaters will become unstable if inadequate 1's density is not maintained. BOC ‐ Bell Operating Company One of the regional telephone companies that were owned by AT&T before divestiture in 1984 (i.e. New England Telephone, Ohio Bell, etc). The 22 BOCs were divided among the RBOCs at divestiture. See RBOC. Both Way Trunks see combination trunks. BRI refers to ISDN Basic Rate Interface ‐ The common form of ISDN with 2 “Bearer” Channels and one “D” channel. All three channels are on a single copper pair and encoded with type 2B1Q coding. BRITE‐ Basic Rate Interface Transmission Extension. A technology where ordinary T‐ 1 trunks (or any other digital carrier system) are used to extend ISDN BRI service. BRITE uses 3 DS0 channel per BRI ‐ One for each B channel and one for the D channel. See Repeater. BTN See Billing Telephone Number. Business Office The part of the phone company where you call if they mess up your bill, to report problems, and to order service. Not necessarily technically literate. Called Party Address This is the destination phone number of a call delivered to a switch. For instance this could be the CLID of a call delivered to a PBX using DID or two‐ way trunks. See also DID. Caller ID A CLASS feature on an analog line that provides the number of the calling line as a ...
Page 359 USER’S MANUAL Calling Line ID See CLID. See also Caller ID. Calling Party Control See CPC. CAS Channel Associated Signaling. A bit‐ based signaling method used on digital lines (such as T1) that is periodically inserted into the low order bit also used for the audio transmission. See Robbed Bit Signaling. Cause Code A code returned by switching equipment to ISDN equipment as part of the call control signaling protocol. The cause code indicates that a network call‐ related event has occurred or has failed. Since these codes actually come from the network, the fact you are getting a cause code is an indication that that the ISDN circuit is operational at some level. CCIS Common Channel Interoffice Signaling. A signaling system where network information such as address and routing information are handled externally to the actual communications (voice) path. SS7 (Signaling System 7) is the internationally standardized CCIS system. Deployment of CCIS increased efficiency since no communications (voice) channels are used merely to report an “all trunks busy” or “far end busy” conditions. It also decreased toll fraud substantially since it removed the potential for access to the signaling information that was inherent to in‐ band signaling schemes. CCIS also enables CLASS features as well as sophisticated re‐ routing features for “intelligent network” applications. See also in‐ band signaling. See also SS7. Central Office See CO Centrex ‐ Central Exchange Service. An enhanced business telephone service intended to offer most of the features of a PBX but where the lines are all from the LEC out of a public switch. Offers CLASS‐ like features for business users such as 4‐ digit "inside" dialing, hold, transfer, attendant, etc. CEPT‐ Conference on European Posts & Telecommunications. This is a European standards body that formerly set the standards for telephone interfaces for 26 countries. CEPT Format The usual rate and frame format for E1 circuits. 2.048 mbps. See E1. CEPT Rate See CEPT format. See also E1. Channel An actual path you can talk or send data over. This is what you are paying the phone company for. For instance, ISDN BRI lines can be ordered with 1 or 2 active channels and these channels can be configured for voice calls (CSV), data calls (CSD) or both (alternate CSD/CSV). A channel does not necessarily have it’s own unique telephone number. See ISDN. Channel Associated Signaling See CAS. See also Robbed Bit Signaling. Choke Exchange A telephone exchange, which is assigned to Radio and TV stations, Promoters, and other users that will be receiving large numbers of simultaneous calls. The idea is to group all of these users on a single exchange so when all routes into that exchange are in use “normal” users (on other exchanges) will not experience blocking of incoming or outgoing calls. Trunks from other local exchanges into the choke exchange are deliberately limited to just a few paths so callers will get an "all trunks busy" instead of completely blocking their local exchange. However, when one of the choke exchange users experiences a large number of calls (as when ...
Page 360 USER’S MANUAL a station contest should not occur as the busy status in response to a call attempt is conveyed over the separate SS7 network. Therefore, the need for choke exchanges has pretty much disappeared. Nonetheless, many Telcos still insist that Broadcasters use special choke lines for call‐ in lines. See blocking and concentration. Circuit A physical path through which electrical signals can pass. It consists of a network of conductors and other components, separated by insulators. Technically this term cannot be applied to fiber optic or other “non‐metallic” paths. See also channel. Circuit Switched Data See CSD Circuit Switching A system where a dedicated channel is allocated to the users of that call for the duration of that call. That channel is allocated for the duration of the call regardless if information is being transmitted at any given moment. Bandwidth through the channel is fixed, at no time may this bandwidth be exceeded. If this bandwidth is not used it is wasted. While inherently inefficient, the dependable and reliable nature of circuit switching makes it ideally suited to real‐ time voice and audio/video conferencing applications. When over loaded Circuit Switched networks will respond "all circuits are busy… try again later". This is in stark contrast packet switched networks or to systems where statistical multiplexing is used. See statistical multiplexing and Packet Switching. CLASS‐ Custom Local Area Signaling Services. A variety of enhanced features (usually on analog lines) that take advantage of the ability of modern SS7 technology’s ability to transmit information about the calling party. CLASS includes such features as Caller ID, Automatic Callback, Call Trace (initiated by subscriber), Selective Call Screening, etc. Clear Cause See Cause Code CLEC‐ Competitive Local Exchange Carrier Your local telephone service provider who is one of the new‐ generation providers rather than a RBOC or Independent. A CLEC is really just an independent, albeit one formed after the divestiture of AT&T. See LEC and Independent. CLI ‐ Calling Line Identity European term for CLID. See CLID. CLID‐ Calling Line Identification This is the ISDN and SS7 equivalent of Caller ID; I.E. the number of the calling party. See also Caller ID and ANI. CO‐ Central Office The Telco facility to which your local telephone circuit lead. Contains “Switches” and “Trunks” as well as the local telephone circuits. Codec‐ COder/DECoder A device which takes digitized audio and “codes” it in order to reduce the transmission bit rate and which can also simultaneously “decode” such coded audio. Strictly speaking, a codec does not include an ISDN terminal adapter and related equipment. Simple codecs are also used in digital telephony. These use a simple companding scheme to reduce channel noise. COL ‐ COnnected Line number European Term. The number to which you have connected. This may not be the number you dialed if call forwarding is used. Combination Trunk A trunk (channel) which can both make and receive calls. This generally refers to analog ground start or loop start trunks, although the term can be applied to ISDN BRI ...
Page 361 Concentration The basic premise is to share facilities wherever possible. For instance, while there may be thousands of customers served by a given Central Office, there will be substantially less than that number of calls which can be handled simultaneously. And, even fewer long distance calls can be made simultaneously. The art of Traffic Engineering is to have enough capacity that calls are rarely blocked, but no more than that number. See also Choke Exchange and Blocking. CPC‐ Calling Party Control Sometimes referred to as "CPC Wink" or "disconnect supervision". A call supervision feature on an analog loop start line that provides the ability for a CO (Central Office) to signal the called party when the calling party hangs up. CPC allows the PBX, key system, or telephone answering device to reset the line so that it is ready to accept or initiate another call. CPC is accomplished by either a loop current drop or reversal. With some CO equipment, it is also provided if the called party drops the call. See also MCLD. CPE‐ Customer Premise Equipment Customer owned equipment located at his/her facility, such as a CSU or terminal. In the USA and Canada, the NT1 is part of the CPE. CPN ‐ Called Party Number European Term. The number that has been dialed. See Called Party Address. CSD‐ Circuit Switched Data A dial‐ up data communications channel which, once established, looks like a transparent data pipe. Also, the type of ISDN service required to utilize this capability of an ISDN circuit. In contrast to CSV. CSU‐ Channel Service Unit The NCTE used in the USA & Canada to terminate a T1 line. Typically the CSU must be provided by the end user. See NCTE and DSX1 CSU/DSU A device which incorporates the functions of a CSU (Channel Service Unit) and a DSU (Data Service Unit) Most commonly it interfaces between a Switched‐ 56 or Dedicated Digital Service circuit and a user’s data equipment such as the Zephyr. CSV Circuit Switched Voice‐ A dial‐ up communications circuit for voice grade communication. Also, the type of ISDN service required to use this capability of an ISDN circuit. In contrast to CSD Custom ISDN (USA & Canada) An ISDN protocol which pre‐ dates National ISDN‐ 1. In most cases National ISDN‐ 1 is also available. The Northern Telecom DMS‐100 switch Supports “DMS Custom Functional" ISDN. The AT&T/Lucent 5ESS switch Supports “Custom Point‐ to‐ Point” (PTP) and Custom Point‐ to‐MultiPoint (PMP). The ISDN protocol has no relation to where one may call. Telos equipment does not support PMP. D Channel‐ Data/Delta Channel Depending on who you ask, it is Data or Delta. The channel which handles ISDN network‐ related data between the user’s equipment and the Telco switch. Used to carry data to set up calls and receive calls. Some Telco’s also allow users to use the D channel to access the packet data network, with appropriate terminal equipment. D4 See Superframe. See also Line Format. Appendix 10 – Glossary...
Page 362 USER’S MANUAL DCE‐ Data Communication Equipment When using serial communications such RS‐ 232, V.35, or X.21, the DCE is the device sending/receiving from the Telco line. i.e.: a modem or CSU/DSU. In contrast to DTE. DDS‐ Digital Data System See Dedicated Digital Service. Dedicated Circuit A permanent channel between two locations. As opposed to a Switched Circuit. Dedicated Digital Service A “Hardwired” or “Nailed Up” digital circuit which is permanently connected between 2 points. Typically 56Kbps or 64Kbps. Dedicated digital lines are frequently cheaper than ISDN for full time service. Also called Digital Data System or DDS. DID‐ Direct Inward Dialing The ability for an outside caller to dial to a PBX extension without going through an attendant or auto‐ attendant. See also DID Number and DID Trunk. DID Extension or DID station A specific phone within a PBX which can be called from the public telephone network without going through an attendant or auto‐ attendant. DID Number A phone number used to route calls from the telephone network to a specific phone in a PBX (the DID extension). DID requires special DID trunks or ISDN PRI “two‐way DID” trunks. Blocks of DID numbers (typically 10 or 20, sometimes higher) are purchased from the LEC or CLEC for use on the PBX. The number of DID numbers usually substantially exceeds the number of trunks in the system. DID Trunk‐ A Direct Inward Dialing Trunk. A trunk (channel) which can only receive calls. A group of telephone numbers (DID numbers) are associated with a given trunk group, however there is no one‐ to‐ one correspondence between the individual channels and these numbers. The PBX uses the DID number given it by the phone company to route the channel to the correct DID extension within the PBX. This allows some or all PBX stations to receive calls directly without going through an attendant (or auto attendant) Note that there are usually more DID numbers than there are DID trunks. See DID number and DID extension. Direct Inward Dialing See DID Directory Number (USA & Canada) Your seven digit telephone number (without the area code), as found in the telephone directory. Directory 1&2 (Zephyr) The Utility menu on the Zephyr where the 7 digit Directory Numbers can be entered during set up. The Zephyr generally does not require these numbers. DN See Directory Number. See also Directory 1 &2 and MSN/DN 1 &2. DNIS‐ Dialed Number Identification Service A service, typically offered by a long distance company on 800 lines, that provides the number dialed by the caller. This allows a caller to receive specific treatment depending on the number dialed. DP Dial Pulse. A method off sending address information by either 1) Causing brief interruptions in loop current. or 2) Causing brief changes of state of a bit on a digital circuit using Channel Associated Signaling. In other words, "rotary" or "pulse" dialing. See also DTMF and CAS. DSØ‐ Digital Signal Level Zero The smallest unit of measure of the standard rate hierarchy used ...
Page 363 USER’S MANUAL DS1‐ Digital Signal Level 1 The second level up the digital rate hierarchy used by the Telcos. This is 24 times the DSØ rate for a total of 1.544 mbps. See DSØ. See also T1. DS2‐ Digital Signal Level 2 Data rate of 6.312 mbps (4 times the DS1 rate). See DSØ and DS1. DS3‐ Digital Signal Level 3 Data rate of 43.232 mbps (28 times the DS1 rate). See DSØ and DS1. DSL‐ Digital Subscriber Line Traditionally refers to an ISDN circuit or sometimes a T1 line, although the term is also frequently used to mean the next generation beyond ISDN. Sometimes xDSL is used to indicate that the writer is referring to any of a number of emerging DSL technologies. DSU‐ Data Service Unit See CSU/DSU. DSX‐ 1‐ Digital Cross Connect level 1 (USA & Canada, primarily) Defined as part of the DS1 (T1) specification and is a closely related signal. The type of signal switched by a Digital Cross‐ Connect System (DACS). The FDL is stripped off at the DACS interface. DSX‐ 1 is also the type of signal that arrives at the user side of a CSU on a T1 line. A DSX‐ 1 cable is limited to 655 feet (200 meters). DTE‐ Data Terminal Equipment When using serial communications such RS‐ 232, V.35, or X.21, the DTE is the device sending/receiving from a modem or CSU/DSU. In contrast to DCE. DTMF – Dual Tone Multi Frequency The standard tone‐ pairs used on telephone terminals for dialing using in‐ band signaling. The standards define 16 tone‐ pairs (0‐ 9, #, * and A‐ F) although most terminals support only 12 of them (0‐ 9, * and #). These are also sometimes referred to as “Touch Tones”. Note that while digital data terminals have the same symbols, ISDN uses “common channel signaling” (over the D channel) and therefore does not necessarily generate any tones at all. However many terminals still generate the tones since they will still be used on occasion to access services (such as voicemail or automated attendant) at the far end using in‐ band tones. E1 A common type of digital telephone trunk widely deployed outside the US and Canada. Has 31 available 64 kbps channels (called DS ’s) plus a sync/control channel for a total data rate of 2.048 mbps. E‐ 1 See E1 ESF‐ Extended Superframe A type of Line format supported on T1 circuits. The Telco determines the line format and line encoding of your line. See Line Format. ETS 300 The pan European ISDN protocol standardized by ETSI. This protocol is used throughout Europe and has been adopted in many other countries outside the USA & Canada. See also MSN. ETSI ‐ European Telecommunications Standards Institute Euro ISDN See ETS 300. Exchange Another name for a Central Office (most often used in European countries). Also used in the USA & Canada to refer to a particular 3‐ digit prefix of a 7‐ digit telephone number. See CO. Extended Superframe See ESF. ...
Page 364 Glare On a POTS line an incoming call is signaled by periodically applying an AC ring voltage to the line. Since there is a semi random period before the ring, and pauses between rings, it is possible to seize a line which is “about to ring” (and answer a call) when attempting to place an outgoing call. When this scenario happens it is called glare. Glare is much less likely if Ground Start or ISDN trunks are used. See also Ground Start Trunk. GR‐ 303 See SLC‐ 96 Grade of service This is simply the ratio of calls blocked to total calls in a decimal form. Therefore, a grade of service of P.08 would represent 8% blocking. Telephone tariffs regulate the acceptable average grade of service which must be provided on public networks. See also Blocking. Ground Start Trunk A type of telephone trunk where the request to make an outgoing call (i.e. request for dial tone) is made by briefly grounding the Tip conductor. Many PBX system use ground start trunks as they are less prone to glare than Loop start trunks. Ground start lines are sometimes used with equipment designed for Loop Start lines. This may or may not work ‐ generally it serves to prevent outgoing calls while incoming calls work normally. Telcos may call these “ground start lines”. See Loop Start Trunk. See also Glare. HDB3‐ High Density Bipolar 3 An E1 line coding method. This is the more modern line coding method of the two commonly available. See Line Coding, T1. See also AMI and B8ZS Hunt group A group of telephone channels configured so that if the first is busy (engaged) the call goes to the next channel, if that channel is busy it goes to the next channel, etc. Hunt groups may hunt from the highest to the lowest, the lowest to the highest, or on some other arbitrary pattern. But the order of hunting will usually be fixed, beginning with one channel and working through (“hunting”) until an unused channel is found. The term may have originated back in the old manual switchboard days when the operator literally hunted for an unused jack to plug a cord into. This arrangement is very common in business scenarios where a single incoming number (the Listed Directory Number) is given to the public, but multiple incoming channels are supported. See also LDN. Hybrid A device which converts from a two‐ wire signal such as POTS lines (or a 2‐ wire intercom) to a four‐wire system (separate send and receive paths) such as used in the pro‐ audio world. While this task is theoretically quite simple, the fact the impedance of most phone lines varies widely across frequency complicates matters. The Telos 10 telephone system was the first practical DSP based hybrid and applied the then brand‐ new technology to this problem. IEC ‐ 1) InterExchange Carrier “Long Distance” carrier. Handles Interlata and interstate calls. Also referred to as IXC. 2) International Electrotechnical Committee A European standards body best known for the power plug now used throughout the world for AC power cords for use on office equipment and computers. Appendix 10 – Glossary...
Page 365 IMUX See Inverse Multiplexing In Band Signaling A signaling system where network information such as address and routing information are handled over the communications (voice) path itself. Usually the information is represented in the form of audible tones, however DC loop current signaling also qualifies as In Band Signaling. See also CCIS. Incumbent Local Exchange Carrier See ILEC. See also CLEC & LEC Independent Any of the phone companies in existence at the time of divestiture that were not affiliated with the Bell System. See RBOC, LEC, and CLEC. Indi See Independent. Interconnect Company A vendor of telecommunications CPE other than a BOC or AT&T. This term was originated by AT&T and was meant to be derisive towards the fledgling industry when the courts said it was OK for end users to buy equipment from someone other than the Bell System. This industry flourished, in spite of AT&T’s disdain, and ironically, the RBOCs were not allowed to sell CPE under the terms of the breakup of AT&T. With the current state of deregulation, the RBOCs are slowly re‐entering this business. Interexchange Carrier See IEC Interwork The ability of two different type of networks to communicate seamlessly. For example, ISDN can interwork calls to both the POTS network and the Switched‐ 56 network. Inverse multiplexing A method of breaking apart and later combining 2 data streams that does not involve the telephone network. Inverse multiplexing is the most common way of achieving this function in Codecs operating at 112 or 128kbps over ISDN. IOC Capability Packages ISDN Ordering Code system. This system was devised by the National ISDN User’s Forum and Bellcore to simplify ordering new ISDN lines in the USA and Canada. Using a single code specifies all line specifications. The Telos Zephyr, ZephyrExpress, TWO, and TWOx12 support IOC package “S”. Search for document NIUF 428‐ 94 at NIUF’s web page http://www.niuf.nist.gov/ for more information. ISDN Integrated Services Digital Network‐ A relatively new and highly flexible type of telephone service which allows dialing on digital channels with multiple bi‐ directional “Bearer” channels each with a capacity of 56 or 64 Kbps and a single bi‐ directional “D channel”. See BRI and PRI. ISDN Protocol The “language” used for communication between the Telco’s switch and the customer’s Terminal Adapter. Each ISDN circuit has one protocol, and the protocol has no effect on where or whom one may call. See ETS 300, National ISDN, and Custom ISDN. ISDN 2 A term used in Europe for ISDN BRI. Also called S . Not to be confused with National ISDN‐ 2. See BRI. Appendix 10 – Glossary...
Page 366 USER’S MANUAL ISDN 30 A term used in Europe for ISDN PRI. Also called S2M See PRI. ISG – Incoming Service Grouping A Hunt Group. See Hunt Group IXC‐ IntereXchange Carrier See IEC Kbps KiloBits Per Second. Measure of digital channel capacity. Key Telephone System A system that allows multiple telephones to share multiple pre‐ determined telephone lines. The system provides indicators to allow the user’s to understand the status of each line available on a given phone. In its most basic form it is USER’S MANUAL up to the user to provide the intelligence to select an unused line, or answer a ringing line, for example. The Telos 10, Direct Interface Module, One‐ x‐ Six, TWOx12 and Series 2101can be classified as Key Telephone Systems. See also PBX. LATA‐ Local Access and Transport Area The area within which calls are routed by your Local Exchange Carrier (LEC). Under the divestiture of the Bell System calls going outside of this area must be handled by an Interexchange carrier (IEC). With the latest round of de‐ regulation the usual IEC companies are being allowed to compete in the IntraLATA long distance market and LECs are beginning to be permitted to handle InterLATA calls. LDN‐ Listed Directory Number When a number of Telco channels share the same hunt group, it is customary to give out only one phone number for the group, although generally each channel will have its own number. The number given out is the “Listed Directory Number” since that is the number that would be listed in the Telephone Directory and given to customers. Sometimes called a Pilot Number. See also DN and Hunt Group. LE ‐ Local Exchange. European term for Central Office. See CO. LEC‐ Local Exchange Carrier Your local telephone service provider which is either an RBOC or an Independent. In other words, a traditional phone company. In contrast to CLEC or IEC. Line An electrical connection between a telephone service provider’s switch (LEC or CLEC) and a telephone terminal or Key system. An electrical connection between a telephone service provider’s switch and another switch is technically called a trunk. Note that some type of physical lines offer more than one channel. I.E. a BRI circuit has 2 channels, called B channels. This term is a confusing one, so we try to avoid using it. See Channel. See also Station Line. Line card The circuit in the Telco switch to which your line is connected. On an ISDN circuit the line card performs a role analogous to the NT1 in adapting to and equalizing the circuit to establish OSI Layer 1. Line Coding, T1 The clock signal for T1 is derived at the far end from the data bits themselves. Therefore, T1 lines have certain restrictions as to the data allowed. No more than 15 zeros shall be sent in a row; and average density of 12.5% ones must be maintained. The CSU is responsible to ensure that these requirements are met. The line encoding method, AMI or B8ZS determines exactly how these requirements are met while still allowing recovery of the original data at the far end. Your Telco will determine the method used on a specific circuit. B8ZS is preferred. E1 circuits have similar restrictions. HDB3 is preferred for E1 circuits. ...
Page 367 USER’S MANUAL Line Format, T1 Modern T1 circuits usually use either Superframe (sometimes called SF or D4) or Extended Superframe (sometimes called ESF) line formatting. The type of framing used is determined by your Telco. ESF is preferred. See ESF and SF Line Side This is the side of a central office switch that the subscriber's telephone lines are connected to. The main reason for distinguishing between this and the trunk side is that certain customer related features (Such as CLASS and Centrex features) are inapplicable to trunks. See also Trunk Side. 2) The user side of a PBX. Also called the station side. Line Termination See LT. Livewire This is the trade name used by Axia Audio (a division of Telos‐Systems) to describe the audio streams utilized in their networked audio system. Local Access and Transport Area See LATA. Local Exchange Carrier See LEC and CLEC. Long Distance If your local Telco is a former Bell Operating Company then any call outside of your LATA or any Interstate call is considered long distance and is handled by an IEC. The above is true regardless of whether you are referring to a dedicated line or a dial up call. 2) However, under the current state of deregulation, toll calls within a LATA may now be covered by the IEC, and in some cases RBOCs are being permitted to handle InterLATA calls. Loop The telephone circuit from the CO to the customer’s premises. Generally refers to a copper cable circuit. Loop Current Disconnect Supervision Another name for CPC. See CPC Loop Qualification Process of actually measuring the loss on a prospective ISDN line to see if it can be used for ISDN service. The actual loss on the line (usually measured at 40 kHz) is the determining factor whether ISDN service can be offered without a repeater. Generally ISDN is available up to 18,000 feet from the serving Central Office. It may not be available within this range, or may be available further from the CO. Only a loop qualification can tell for sure. Not all Telcos will extend ISDN lines with repeaters. Loop Start Line A plain old telephone line. The telephone terminal signals the “off hook” condition by allowing DC current to flow. See Ground Start Trunk. See also Glare. Loop Start Trunk A plain old telephone line connected to a PBX switch. See Loop Start Line. The PBX signals the “off hook” condition by allowing DC current to flow. Ground Start Trunks are generally preferred for use on PBXs to prevent glare. See Ground Start Trunk. See also Glare. The Telco may call this a “Loop Start Line”. LOS Loss Of Signal. An LED or other indicator that illuminates if a signal is absent. This terminology is commonly used with T‐ 1 equipment. LT ‐ Line Termination The electrical and protocol specifications for the Central Office end of an ISDN line. If you wish to connect an ISDN terminal (such as a Zephyr Xstream) to a PBX the PBX must support LT ISDN. See also NT and Line Card ...
Page 368 (Euro & INS 64 ISDN) or 7 digit Directory Numbers (USA ISDN) can be entered during set up. The Zephyr Xstream generally does not require that these numbers be present. See Directory Number and MSN. MSN‐ Multiple Subscriber Number (Euro ISDN) This is a telephone number associated with an ETS 300 BRI line. Providers of ETS 300 often give you three MSNs with a BRI, although additional MSNs can be purchased. An ISDN terminal will “ring” (provide an alerting signal) only when calls are made to the MSN (or MSNs) entered in that terminal. If a terminal has no MSNs entered, it will “ring” whenever there is a call to any of the MSN’s on that BRI. See ETS 300 and DN. National ISDN (USA & Canada) The US “standardized” multi‐ platform ISDN protocol. The first version is National ISDN‐ 1. As of mid 1996 National ISDN‐ 2 has been implemented in some areas and is fully backward compatible with National ISDN‐ 1. NCTE – Network Channel Terminating Equipment. NCTE is a general term that can be applied to a CSU or NT1 or other equipment terminating a digital line at the customer’s premises. In many countries, the NCTE is provided by the Telco. The USA is not one of those countries. Network Channel Terminating Equipment See NCTE. Network Termination See NT NIUF‐ National ISDN User’s Forum A user’s group formed under the National Institute of Technology (NIST) in the USA. The NIUF is a neutral forum where the switch manufacturers and Telcos can get input from users and CPE manufacturers regarding the implementation of National ISDN. Among NIUF’s successful projects have been the IOC ordering codes. Their web page is at http://www.niuf.nist.gov/ . See National ISDN. Nortel Manufacturer of the DMS‐ 100 switch as well as many other pieces of telecom and network equipment. Now called Nortel Networks. Northern Telecom The Canadian company which was once the manufacturing arm of Bell Canada (it was called Northern Electric back then). Now called Nortel Networks. See Nortel. NT ‐ Network Termination The electrical and protocol specifications for the user end of an ISDN line. See also LT. NT‐ 1 An alternative expression for NT1. See NT1 NT1‐ Network Termination Type 1 The termination at the customer premises of an ISDN BRI circuit. The NT1 performs the role of line termination of the “U” interface and Codes/Decodes from the line’s 2B1Q coding scheme. The customer end of the NT1 interfaces using the “S” or “T” interface. The NT1 is frequently part of the “Terminal Adapter” and is built‐ in to Zephyr Xstream, Zephyr, ZephyrExpress, Telos TWO and TWOx12 systems sold in the USA & Canada. See also NCTE NTBA‐ Network Termination Basic Access The term used for NT1 in some countries. See NT1. See also NCTE. Appendix 10 – Glossary...
Page 369 USER’S MANUAL Packet Switching Packet Switched networks are more commonly associated with Computers, Local Area Networks, and the Internet. In a packet switched network the raw stream of data is broken into individual pieces, called packets. Each packet is routed through the data network, individually. This is somewhat analogous to taking the pages of a book and sending each page as a letter through the postal system. The page numbers would allow reassembly of the book no matter what order the pages were received at the far end. The end user does not know or care that the packets may travel a variety of routes. If a given page did not arrive in a reasonable length of time, one could request that this page be re‐ sent. Most packet switched systems allow packets to be discarded if the network capacity is exceeded (the postal system is not supposed to do this). This is accommodated by the higher‐ level protocol, which knows to request that a packet be re‐ sent if it does not arrive. Therefore, the typical behavior of a packet switched network when overloaded is that throughput decreases (i.e. the network "slows down") as the percentage of discarded packets increases. In stark contrast to Circuit Switched networks. See Circuit Switching. PBX‐ Private Branch Exchange A privately owned switch. Basically, a PBX is a private “business” telephone system which also interfaces to the telephone network. In some circles 'PBX' implies a manual switchboard whereas 'PABX' (Private Automatic branch exchange) implies a PBX that supports dialing by end users. Many PBX’s can now offer ISDN BRI service, usually over the S Interface. A few vendors are now offering BRI over the U interface as well. PRI over DSX‐ 1/T1 or E1 is also offered in some cases. Be wary of these ISDN protocols since they have not been as well tested as the versions running on “public” switches. They may or may not work with a given piece of CPE. PIC‐ Primary Interexchange Carrier (USA) This is your default “1+” carrier used for interLATA calls. In some areas you may have two PICs, one for interLATA calls, and one for intraLATA long distance calls (in which case it stands for Primary Intraexchange Carrier). In some areas intraLATA long distance calls are still handled by your RBOC, in others you now have a choice. You may be able to discover who you current PIC is by dialing 700 555‐ 4141. PMP (USA) AT&T “Custom Point to Multi‐ Point” Custom ISDN Protocol. Not supported by the Zephyr or ZephyrExpress. See Custom ISDN and ISDN Protocol. POP ‐ Point Of Presence The local facility where your IEC maintains a switch. This is where your long distance calls get routed so that your IEC can handle them. Also used to describe the local access point of an Internet Service Provider. Port This is a pretty general term. Newton’s Telecom Dictionary 10th edition defines a port as “An entrance to or an exit from a network”. Many phone equipment vendors refer to ports as the physical interface between a Switch and a Line or Trunk. Product literature often refers to the number of ports on a phone system. In this context it refers to the number of phones or lines (or sometimes the combination) the system supports. POT‐ Plain Old Telephone A black, rotary‐ dial desk phone. Usually a Western Electric model 500 set. Outdated term. ...
Page 370 USER’S MANUAL Primary Rate Interface A form of ISDN with 23 “B Channels” and one “D channel”. All 24 channels are on a single cable. Functionally related to T1 telephone circuits. In Europe PRI has 30 “B Channels” and one “D Channel” and one “Sync channel”. See also B channel and D channel. Provisioning The act of configuring an ISDN or other telecommunications path. Also refers to the complete line configuration information. PS2 Power Power provided on pins 7 and 8 of the “S” interface cable. This power is used so that a NT1 can provide power to a terminal (usually a phone). In some cases, it is used to allow a terminal to power an NT1. The USA versions of the Zephyr, ZephyrExpress, supply PS2 power in the “S” jack. The Zephyr Xstream and Telos TWO provide this on all versions. This power arrangement is also used in the Telos TWOx12 and 2101 Studio Interface to power Desktop Directors. PTP (USA) AT&T “Custom Point to Point” Custom ISDN Protocol. Point‐ to‐ Point lines have only one incoming phone number which must be dialed twice to connect to both lines (the first call goes to “line 1” and the second call rolls over to “Line 2”. See Custom ISDN and ISDN Protocol. QoS (Quality of Service) is the ability to provide different priority to different applications, users, or data flows, or to guarantee a certain level of performance to a data flow. For example, a required bit rate, delay, jitter, packet dropping probability and/or bit error rate may be guaranteed. Quality of service guarantees are important if the network capacity is shared, especially for real‐time streaming multimedia applications such as Livewire . RBOC‐ Regional Bell Operating Company (USA) One of the regional companies formed when AT&T was forced out of the local telephone business. Each RBOC (or “baby bell”) owns a number of the former “Bell Operating Companies”. The Bell Operating Companies are the traditional local phone companies (pre‐ 1984), except where one’s service is from an “Independent” (non bell) telephone company or a CLEC. Due to their former association with the Bell System RBOCs are regulated by the FCC differently than are independent Telcos or CLECs. In many cases the Bell Operating Company structure is no longer used. For instance, here in Ohio we now deal directly with the RBOC, Ameritech, while the old Bell Operating Company, Ohio Bell Telephone, no longer exists. Another trend is mergers among the RBOCs (and in some cases the independents as well). See CLEC and LEC. RD‐ Receive Data Data coming from the network, or DCE towards the DTE. Also, a light on a modem or CSU/DSU that lights to indicate presence of this signal. Red Alarm An alarm state on a T‐ carrier circuit that indicates that the incoming signal (at the network interface) has lost frame for more than a few seconds. Normally a Yellow alarm is then returned (i.e. sent back) if a Red alarm is present. A Red Alarm indicates a loss of inbound signal; a Yellow alarm indicates (indirectly) a loss of outbound signal. See also Yellow alarm, Blue alarm, and LOS. Regional Bell Operating Company See RBOC ...
Page 371 USER’S MANUAL Robbed Bit Signaling A signaling scheme that “borrows” bits on each T1 channel for use as signaling channels. On SF T1’s there are two bits, the A bit and the B bit in each direction. On ESF T1’s there is also a C and D bit in each direction, although they are rarely used. Using these bits, various older analog trunk interfaces can be emulated over a T1. For instance, dial pulse address signaling using 10 pulse per second (rotary style) digit groups over these bits. Since robbed bit signaling interferes with the least significant bit, only 7 bits can be used for sensitive data applications, leaving only a 56kbps channel for data applications. See also CAS and CCIS. Rollover See Hunt Group S interface The electrical interface between the NT1 and the Terminal Adapter or other ISDN equipment. ISDN equipment with built‐ in NT1’s do not necessarily provide access to the S interface (the Zephyr Xstream, Zephyr, ZephyrExpress and Telos TWO do). Multiple devices can share an NT1 by connecting on the S interface. Also known as the S passive bus. S European term for ISDN BRI. See BRI and ISDN2. S2M European term for ISDN PRI. See PRI and ISDN 30 Sealing Current Unlike telegraphy, teletypewriter and POTS lines, most digital lines (such as ISDN) use a voltage rather than current mode of operation. Sealing Current allows a controlled amount of current to be passed through a telecom circuit for purposes of “healing” resistive faults caused by corrosion. Bellcore specifies sealing current on the ISDN U interface in the USA. The Siemens EWSD switch does not provide sealing current. Most other ISDN capable switches used in North America do. SF‐ Superframe A type of Line format supported on T1 circuits. The Telco determines the line format and line encoding of your line. ESF is the preferred Line Format on T1 circuits. See Line Format. Silence Suppression See Statistical Multiplexing. SLC‐ 96 A Subscriber Loop Carrier Circuit system manufactured by AT&T (now Lucent). SLC‐ 96 has its own version of T1 framing between it and the CO. SLC‐ 96 and similar "SLIC" systems may or may not perform a concentration function. The interface is the Bellcore TR‐ 008 or the newer GR‐ 303 interfaces that are specialized versions of T1 intended to allow transparent transport of analog CLASS features such as Caller ID, Call Waiting, etc. The GR‐ 303 interface is specifically intended to be used as a common point of interconnection between alternative equipment, technologies, and/or networks (i.e. voice‐ over‐DSL, voice‐ over‐ IP, etc)) and the public switched network. See the following link for additional information from Telcordia: http://www.telcordia.com/resources/genericreq/gr303/index.html SLIC 1) Subscriber Line Interface Circuit, see “Line Card”. 2) The equipment used with the AT&T (Lucent) SLCC Subscriber Loop Carrier Circuit, a system used to multiplex a number of subscriber loops onto a single circuit (usually a T1 circuit) to reduce fixed costs. 3) Also sometimes used generically for other brands of similar equipment. See also SLC‐ 96. ...
Page 372 USER’S MANUAL Custom PMP, and DMS custom ISDN protocols the equipment requires that the user to program SPIDs into it. Custom PTP and ETS 300 protocols do not require a SPID. No SPIDs are required for ISDN PRI protocols. SS7 ‐ Signaling System 7 The internationally adopted Common Channel Interactive Signaling (CCIS) system. Previous to SS7 the Bell System used SS6 which did not support the International Standards. SS7 does. It allows for substantially flexibility and power in dynamically routings calls. An SS7 database lookup is how a call to a mobile telephone user can handled transparently despite the fact that the user's location may change. Also used to determine what carrier should handle a given toll free call. See also CCIS. Station Line A telephone circuit from a PBX to a telephone on that PBX. Since this is a telephone‐ to‐ switch connection it is considered to be a “line”. See Line and Trunk Station Side The user side of a PBX. The side of the switch that the telephones are attached. Also, occasionally called the 'line side'. The main reason for distinguishing between this and the trunk side is that certain customer related features (Such as Hold and Transfer) are inapplicable to trunks. See also Trunk Side. Statistical Multiplexing A method of improving effective bandwidth of a Telco channel. Statistical Multiplexing takes advantage that there are typically many pauses in a conversation. By taking advantage of this fact, and not sending the pauses, improvements in efficiency can be made. Also referred to as silence suppression. See Circuit Switched. Subscriber The customer of a Telecommunications company. This term dates back to when a local Telephone Company was formed at the specific request of a group of customers who agreed in advance to “subscribe” to the service. Superframe See SF Switch Telephone switching device which “makes the connection” when you place a call. Modern switches are specialized computers. ISDN service is provided from a “Digital” switch, most commonly (in the USA & Canada) an AT&T model “5ESS”, Northern Telecom model “DMS‐ 100”, or Siemens model “EWSD”. The switch, and related software running on it, will determine which ISDN protocol(s) will be available to customers connected to it. See also PBX. Switch Type (ZephyrExpress) The utility menu item where the ISDN Protocol is selected. See Telco Setting and National ISDN. Switched Circuit A channel which is not permanent in nature, but is connected through a switching device of some kind. The switching device allows a switched circuit to access many other switched circuits (the usual “dial up” type of telephone channels). Once the connection is made however, the complete capacity of the channel is available for use. As opposed to a dedicated circuit or a packet based connection. Switched‐ 56 A type of digital telephone service developed in the mid 1980’s which allows dialing on a single 56Kbps line. Each Switched‐ 56 circuit has 1 or 2 copper wire‐pairs associated with it. You will need to know in advance if you will have the two‐ wire or four‐wire version so you can procure the appropriate CSU/DSU. Switched‐ 56 is being rapidly replaced with ISDN, ...
Page 373 USER’S MANUAL Synchronous Data A form of serial data which uses a clock signal to synchronize the bit stream. Since, unlike asynchronous data, no start and stop bits are used, data throughput is higher than with asynchronous data. ISDN and T‐ 1 use Synchronous data. See also Asynchronous Data. T‐ Link A proprietary Telos 2.048mbps channelized link. This link uses the DSX‐ 1 electrical protocol and has 30 channels at 64kbps each. T1 A common type of digital telephone carrier widely deployed within the US, Canada, and Japan. Has 24 64Kbps channels (called DS ’s). The most common framing scheme for T1 “robs” bits for signaling leaving 56kbps per channel available for data use. T‐1 An alternative expression for T1. See T1. TA ‐ Terminal Adaptor. The electronic interface between an ISDN device and the NT1. The terminal adapter handles the dialing functions and interfaces to the user’s data equipment as well as to the NT1 on the “S” or “T” interface. Tandem Switch A switch which is between two others. It connects two trunks together. Long distance calls on a LEC line go through a long distance tandem that passes them through to the long distance provider’s switch. Tandem Tie Trunk Switching When a PBX switch allows a tie line call to dial out of the switch. For example, if switch “A” in Arkansas has a tie line to switch “B” in Boise, Boise could use the tie line to make calls from switch “A”. TD‐ Transmit Data Data coming from the DTE towards the DCE or network. Also, a light on a modem or CSU/DSU that lights to indicate presence of this signal. TEI ‐ Terminal Endpoint Identifier An OSI Layer 2 identifier used by an ISDN terminal to communicate with the serving CO. Most equipment uses a dynamic TEI assignmment procedure . Telco‐ Telephone Company Your local telephone service provider. In the 21st century you generally have a choice of Telcos if you are a business in a major metropolitan area in the USA. Competition is coming to the Telecom industry around the world. Telco Setting (Xstream, Zephyr, TWO, TWOx12) The menu selection where the ISDN protocol is selected. Choices are 'Natl I‐ 1' (for National ISDN), 'DMS Cust', (for DMS‐100 Custom Functional ISDN), 'AT&T Cust' or 'PTP' (for AT&T Custom Point‐ to‐ Point), 'INS 64' (for Japanese ISDN) and 'ETS300' (for Euro‐ ISDN). Telcordia Technologies Formerly Bellcore. The research and development organization owned by the telephone companies. Telcordia represents the phone companies in developing standards for Telco equipment and in testing equipment compliance to those standards. Promotes competition and compatibility through standards promoting interoperability such as GR‐ 303. Telcordia also offers educational and training programs open to all interested parties. Bellcore was sold to SAIC in 1997. Telcordia is responsive to both RBOCs and independent Telcos. Their web site is: http://www.telcordia.com . See GR‐ 303 ...
Page 374 USER’S MANUAL Telos Test Line (Xstream, Zephyr & ZephyrExpress) 24 Hour test line which can be called to verify correct operation of your Zephyr or ZephyrExpress codec. Numbers are +1 216‐ 781‐ 9310 and +1 216‐ 781‐ 9311. The test line transmits in dual/mono mode at 32kHz and can be received in L3 Mono using one B channel or L3 Stereo using 2 B channels. Telos also maintains an identically configured test line in Freising, Germany at 49 81 61 42 061 (dial this number twice). Terminal Adapter See TA Tie Line See Tie Trunk. Tie trunk A Trunk between two PBXs. Note, a tie line is a dedicated circuit, not a switched circuit. See Trunk. TR‐ 008 See SLC‐ 96 Trouble Ticket A Telco “work order” used to track Customer Repairs within the Telco. If you call someone “inside” the Telco’s repair department, they will need this number to proceed. It will also be needed whenever you call to check on the status of a repair. Always ask for this number when initiating a repair request. Trunk A communications path between two switching systems. Note that many trunks may be on a single circuit (if that circuit has multiple channels). The trunks most users will deal with are between the Telco switch and a PBX. However, a Tie Trunk can connect two PBXs. See also Tie Trunk and Trunk Group Trunk Group A number of telephone channels which are functionally related. Most common is the Hunt Group. Other common types include Incoming Trunk Groups and Outgoing Trunk Groups. See also Combination Trunks and Two‐ way DID Trunks. Trunk Side 1) The side of a central office that faces the network. Historically many CO switches could not make trunk to trunk connections (as opposed to tandem and long distance switches, that are always used to connect trunk together). Hence the need to distinguish between the "line side" and the "trunk side" of the switch. See also Line Side. 2) The side of a PBX that connects to the Telco. Historically many PBXs could not make trunk to trunk connections. Hence the need to distinguish between the "line side" and the "trunk side" of the switch. Since a trunk is a switch‐ to‐ switch circuit, these circuits can be called trunks. Beware, even though you have a PBX, the Telco may still call these 'lines' (even though your PBX considers them trunks). See also Trunk and Line Side Two‐ way DID trunk An ISDN PRI equipped for direct inward dialing. Most PBX trunks are related to a given phone number, either alone or as part of a hunt group. In the case of a “normal” (i.e. analog) DID Trunk, a group of phone numbers are associated with that DID trunk (or group of trunks) and incoming calls include the DID number, so the PBX can route that call to the correct DID extension. These are one‐ way (i.e. inward only) trunks. This is exactly how ISDN PRI functions, with the DID information coming in over the D channel. There is a big difference between a normal DID Trunk and a Two‐ way DID trunk over ISDN PRI. For one thing, ISDN PRI is digital. Another distinction is that you cannot dial out over a true DID trunk, while you can dial ...
Page 375 USER’S MANUAL U Interface The interface between the ISDN “BRI” line and the “NT1”. This can be considered the ISDN “phone jack” in the USA & Canada, and is frequently in the form of a RJ‐ 11 or RJ‐ 45 style telephone jack. The center 2 pins are used on these jacks. V.35 A serial data interface/connector pin out for synchronous data. V.35 uses balanced signal and data lines. The Zephyr/Zephyr Xstream models 9X01, 9X00, support V.35 using part #9812 cable. Variant The particular protocol (i.e. National ISDN‐ 1 or ETS 300) running on a specific switch. Not all variants are valid for a specific switch. The switch brand and model plus the variant defines the ISND protocol. Applies to configuring the 2101. See ISDN Protocol. Virtual ISDN An alternative to repeaters which uses a local Telco Switch to act as a repeater and which then sends the signal onto another switch which supports ISDN. See also Repeater. Work Order See Trouble Ticket X.21 A serial data interface for synchronous data popular in Europe. X.21 uses balanced data and unbalanced signal lines. The Zephyr models 9X01, and 9X00 support X.21 using part #9822 cable. Yellow Alarm An alarm on a T‐ carrier circuit that is returned by the local equipment if it is in a Red Alarm state. A Red Alarm indicates a loss of inbound signal; a Yellow alarm indicates (indirectly) a loss of outbound signal. See Red Alarm, Blue Alarm and LOS. Appendix 10 – Glossary...
Page 376 USER’S MANUAL Appendix 10 – Glossary...
Page 377: Index
CODEC Menu Detailed, 203 CODEC MENU In Netcoder Mode, 317, 318 Codec Settings: Transmit, Receive, Bit Rate, & Sample Rate, 78 Codecs Connecting to non-Telos Codecs. See Coding Method vs Fidelity Quick Comparison, 172 coding methods, 111 Cold Boot. See Restore Factory Defaults Command Line Interface Ethernet, 261 RS‐232, 249...
Page 378: Environmental Considerations
Detailed, 227 Guided Tour, 105 Dial Setups Activating a Dial Setup – Step by Step, 70 Creating Dial Setups – Step by step, 71 Editing, 72 Triggered by Closures, 244 When INterface is set to V.35, 244 Digifon, 301 Digital Data System Using with V.35/X.21 Interface, 158 Directory Numbers Detailed, 221...
Page 379 masking, 112 Menu System (introduction), 61 Menu Tree, 83 Menus Detailed Reference, 189 Guided Tour, 83 Meters and indicators, 40 Mix-Minus, 186 Mono‐128, 124 MPEG, 116 MPEG LAYER 2, 124 multi-band compression, 114 Multiple Codec Connections, 178 multiple codecs, 126 Multiple Subscriber numbers Detailed, 221 Multiple Subscriber Numbers Guided Tour), 103...
Page 380 Guided Tour, 99 Sample Rate (introduction), 80 SEDAT, 126, 127 Send Bargraph, 20 Setting the nominal input level range (non-mixer only), 264, 265 Software Updates, 267 Special Menus; AUTO, DIAL, & DROP, 66 Special Operational Modes Ethernet Interface, 317 V.35 Single (high speed), 325 SPECIFICATIONS, 295 SPID Error, 217, 286 SPIDs...
Page 381 Installation, 155 Voice-over applications, 174 Volume, 230 Volume (front panel), 108 WARRANTY, 295, 298 Web Browser Control via, 261 Zephyr Remote, 262 Index...

Telos Zephyr Xstream User Manual

1 QUICK START - Step by Step Guide to Getting Running

2 INTRODUCTION - Getting to Know the Zephyr Xstream

3 GUIDED TOUR of the HARDWARE

4 Basic Operation & Installation

5 GUIDED TOUR of the MENUS

6 Audio Coding Reference

7 Isdn Basics

8 LIVEWIRE -IpAudio

LIVEWIRE TM - IP Audio ............................................................................................................... 143

9 The V.35/X.21 Interface Option

10 APPLICATIONS GUIDE - Hints & Ideas

11 THE WORKS! - Detailed Menu Reference

12 System Functions and Remote Control

13 Deep Tech Information & Troubleshooting

14 Specifications & Warranty

Appendices

Appendix 1: Codec Interoperability Information

Appendix 2: Finding Public Codec User Lists

Appendix 3: ISDN Cause Phrases/Values

Appendix 4: Known Working SPID Formats by Telco

Appendix 5: Suggested Reading and References

Appendix 6: Streaming and Special Operational Modes

Appendix 7 - Ordering ISDN for Countries Using ETS300 (EURO ISDN)

Appendix 8: Ordering ISDN for the USA & Canada

Appendix 9: Modular Cable Guide

Appendix 10: Glossary

Index

Quick Links

Troubleshooting

Related Manuals for Telos Zephyr Xstream

Summary of Contents for Telos Zephyr Xstream

Table of Contents