PERMANENT DAMAGE MAY OCCUR! If you are unfamiliar with any facility, check that the line you are using is NOT a digital line. If the Tieline codec becomes faulty due to the use of a digital phone system, the WARRANTY WILL BE VOID.
· Other popular algorithms including LC-AAC, HE-AAC v1 and v2, AAC-LD, AAC-ELD, AAC-ELDv2, Opus, MPEG-1 Layer II and III, Tieline Music and MusicPLUS, G.722 and G.711. · SmartStream PLUS redundant streaming for high reliability over IP networks without Quality of Service.
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Merlin PLUS User Manual v1.4 E.g. Session Type and Data are not displayed when configuring Sessionless IP connections. · Default settings may also change depending on the session type selected, e.g. Tieline Session versus SIP or Sessionless. ISDN Menu Navigation Select Connect >...
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5. Navigate to the preferred setting and Configure Audio Reference Metering when Connecting to Tieline G3 Codecs New generation Genie, Merlin and Bridge-IT IP codecs have more audio headroom than Tieline G3 audio codecs, therefore the audio metering reference scale needs to be adjusted when connecting to a Commander or i-Mix G3 codec with one of these codecs.
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(Note: this is the same as the AES Rx Clock setting in Tieline G3 codecs). The codec will initially try to use the signal on AES inputs 1 and 2 as the clock to which the AES outputs are synchronised. If unavailable, it will then attempt to use inputs 3 and 4 or inputs 5 and 6 in that order.
The NT-1 is a relatively simple device that converts a 2-wire U interface into the 4-wire S/T interface. If you have an NT-1 device connected to the U interface line then you will require a Tieline Euro ISDN G5 module (S/T interface - model: TLISDNEUROG5). If you don’t have an NT-1 device installed then the Tieline US ISDN G5 module (U interface - model: TLISDNUSG5) will be required.
· Expected dialing behaviors, e.g. if B channels should bond or not, and whether audio streams need to use Route tags. · The type of call being made, e.g. Tieline (with Tieline Session Data) versus non-Tieline (sessionless calls). Adjust answering configuration via Settings > Answering > ISDN Answer Configs > [Select...
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Tieline session data from the caller and configure its own algorithm settings according to that. If it fails to receive Tieline session data within 5 seconds (i.e. a non-Tieline codec is calling, or a Tieline codec with session data disabled), it will use the settings in the ISDN Answering Config instead.
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Merlin PLUS User Manual v1.4 8. Select the default algorithm when receiving a call from a non-Tieline codec, then press the button. 9. Specify the audio stream Route when receiving a call on the answering codec from a non-Tieline codec, then press the button.
POTS connections. Connecting to G3 Codecs using POTS The codec will successfully connect to Tieline Commander G3 and i-Mix G3 codecs over POTS. These Tieline G3 codecs may use: · POTS modules (older superseded version) ·...
All POTS G5 modules are capable of making analog voice calls. It may be necessary to make an analog call to dial a telephone hybrid, or to use for communications, or because there is no Tieline codec at the other end of the link. Remember analog voice calls are only 3 kHz audio quality. To select analog phone answering mode in a POTS G5 module navigate to Settings >...
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Calls are answered based on the settings in Config 1 & 2 via Settings > Answering > POTS Answer Configs. Adjustments to these Config settings are not normally necessary when connecting between Tieline codecs. Default settings may need to be adjusted when connecting to non-Tieline codecs over POTS (see POTS Answering Configuration for more info).
Connection setting preferences are normally exchanged via session data sent between two Tieline codecs when a connection is established. If you answer a call from a non-Tieline codec you will need to create an answering "Config" to determine the settings used when connecting, and designate which module will answer the call (if more than one POTS module is installed).
Important Note: On the Comrex codec select its "Music" algorithm. Please note that 9.6kbps connections are not supported by the Comrex codecs. 7. If required you can specify the audio stream Route when answering a call from a non-Tieline codec, then press the button.
Tieline Genie and Merlin codecs use programs to connect to another codec. A Program configures a Tieline codec to send or receive one or more Audio Streams based upon the particular application the codec is being used for at any given time. The attributes of the audio stream and associated connections are embodied within a program when it is created, including the configuration, dialing and answering parameters.
2. Press the HOME button to return to the Home screen, select Connect > IP > Tieline and press the button. Note: Select SIP or Sessionless instead of Tieline if these connections are required.
20.3 Steps to Connect over ISDN The following procedure explains how to create a custom peer-to-peer program and dial another Tieline codec over ISDN using the front panel keypad and navigation buttons. Important Notes: · See Testing ISDN Connections for valuable information about setting up and maintaining reliable ISDN connections.
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3. Navigate to Setup and press the button. 4. Select whether to dial with Tieline Session Data or select Sessionless if dialing a non-Tieline codec, then press the button. Important Note: By default, when Tieline codecs dial they send call configuration settings to the remote codec using Tieline Session Data.
Steps to Connect over POTS 20.5 The following procedure explains how to create a custom peer-to-peer program and dial another Tieline codec over POTS using the front panel keypad and navigation buttons. Important Notes: · See POTS Connection Tips and Precautions for valuable information about setting up and maintaining reliable POTS connections.
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7. Select an algorithm, then press the button. 8. Select Tieline Codecs session data when connecting to another Tieline codec or Sessionless when dialing to non-Tieline POTS codecs. Important Notes: To dial a Comrex® Vector, Matrix® or BlueBox® codec over POTS select the Other algorithm and Sessionless.
G3 audio reference level. To do this select SETTINGS > Audio > Ref Level > Tieline G3. In addition, select the following on the G3 codec prior to dialing. · Select either a mono or stereo profile · Select [Menu] > [Configuration] > [IP1 Setup] > [Session Type] > [SIP] ·...
2. Use the navigation buttons to select Connect and press the button. 3. Select IP and press the button. 4. Select Tieline session mode and press the button. Note: algorithm profiles are only available for Tieline session connections. 5. Use the down navigation button to select Setup and press the button.
Merlin PLUS User Manual v1.4 Connecting to the ToolBox Web-GUI There are three graphical user interface (GUI) options for configuring Tieline G5 codecs: 1. Java Toolbox Web-GUI: codecs can be fully configured including program creation, dial and hangup, command and control.
2. When you launch Toolbox an authentication dialog prompts you to enter a password to login. The first time you log in you can enter the default setting "password" and click the OK button. Tieline highly recommends you click the hyperlink in the login dialog or visit...
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1. Click to launch the HTML5 Toolbox Web-GUI. 2. When you launch Toolbox for the first time an authentication dialog prompts you to enter the user name "admin" and password "password" to login, then click the OK button. Tieline highly recommends you change the password (see Changing the Default Password).
The default password for the Toolbox Web-GUI is password. Enter this in the authentication dialog to use the Web-GUI initially and then Tieline highly recommends changing the default password to protect your codec from being tampered with during live broadcasts.
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Merlin PLUS User Manual v1.4 Inputs Panel for Input Adjustments Important Note: Tieline codecs have different input configurations, therefore the image shown may not reflect the number of inputs displayed in your codec Web-GUI. Feature Description Channel ON/OFF Buttons Click to turn each channel ON or OFF...
An IP address is a unique address to identify a device on a TCP/IP network. Your codec uses dual IP protocol stacks to allow your codec to work on both IPv4 and IPv6 networks. Your Tieline codec supports both DHCP (default) IP addressing and static IP addresses for dialing IPv4 connection endpoints.
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· Expected dialing behaviors, e.g. if B channels should bond or not, and whether audio streams need to use Dial and Answer Route tags. · The type of call being received by the codec, e.g. Tieline (with Tieline Session Data) versus non-Tieline sessionless calls.
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Config 1-4 in ISDN Answer. You can also select the default algorithm. For example, if a call from a non-Tieline codec is received via B Channel 1 on Module 1 (i.e. no Dial Route has been specified in the dialing codec): 1.
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If it fails to receive Tieline session data within 5 seconds (i.e. a non- Tieline codec is calling, or a Tieline codec with session data disabled), it will use the settings in the ISDN Answering Config instead.
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Allow Answering of Sessionless ISDN Calls Only Select Sessionless Only when answering ISDN calls from non-Tieline codecs only. When Sessionless Only is selected, the codec will not wait to receive the Tieline session data. This reduces the time taken to answer an inbound sessionless call.
The default Config settings for POTS modules are designed to suit Tieline codecs. These settings will need to be adjusted to connect to non-Tieline POTS codecs or connect in Analog Phone mode.
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Calls are answered based on the POTS Answer settings in Config 1 & 2. Adjustments to these Config settings are not normally necessary when connecting between Tieline codecs. They are normally adjusted when connecting to non-Tieline codecs over POTS (see...
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· Expected dialing behaviors and encoding, e.g. whether audio streams use Route tags and which algorithm is used. If you answer a call from a non-Tieline codec you will need to create an answering "Config" to determine which module in the codec will answer the call and the settings used when connecting.
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POTS connections. When dialing Tieline to Tieline over POTS using the Merlin or Genie family of codecs, you can configure a Dial Route in the dialing codec's program and a corresponding Answer Route in the answering codec's program. This will ensure a particular audio stream is routed between two codecs consistently.
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Config 1 or 2 in POTS Answer. You can also select the default algorithm. For example, if a call from a non-Tieline codec is received via POTS Module 1 (i.e. no Dial Route has been specified in the dialing codec): 1.
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The Return Audio Port is used by the local codec to receive audio from the remote codec. When Tieline Codecs is the Session Protocol selected (using Tieline session data), the default port value for the Return Audio Port is Automatic.
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The time frame against which lost data is measured Keep Alive The keep connection alive time before failing over to a backup connection; Tieline RTP pings every second to confirm connectivity Automatic Resume Select the check-box to configure fail back to a higher...
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FEC settings. Important Note: The Return Audio Port is the port used by the local codec to receive audio from the remote codec. When Tieline Codecs is the Session Protocol selected (using Tieline session data), the Return Audio Port is automatically configured as UDP audio port 9000 by default for the first audio stream connection.
It is possible to create two simultaneous mono peer-to-peer audio stream connections with different codecs. This is similar to a 'Dual Mono' profile in G3 Tieline codecs. The following program wizard procedure displays the configuration screens to create an answering connection for each incoming call.
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4. Enter the name of the connection in the text box, then click Next. 5. Configure the transport settings: For IP click the drop-down Session Protocol menu and select Tieline Codecs and ensure the Any check-box is not selected, then click Next.
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6. Enter the name of the second audio stream connection in the text box and click Next. 7. Configure the transport settings: For IP click the drop-down Session Protocol menu and select Tieline Codecs and ensure the Any check-box is not selected, then click Next.
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Master panel. When this program is loaded any codec dialing in using IP1 (using default Tieline IP port settings) will be routed to output 1 on the codec and a codec dialing in using IP2 will be routed to output 2.
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The Return Audio Port is used by the local codec to receive audio from the remote codec. When Tieline Codecs is the Session Protocol selected (using Tieline session data), the default port value for the Return Audio Port is Automatic.
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The time frame against which lost data is measured Keep Alive The keep connection alive time before failing over to a backup connection; Tieline RTP pings every second to confirm connectivity Automatic Resume Select the check-box to configure fail back to a higher...
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4. Enter the connection name in the text box, then click Next 5. Configure the transport settings: For IP click the drop-down Session Protocol menu and select Tieline Codecs and ensure the Any check-box is not selected, then click Next.
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8. Enter the IFB audio stream connection name in the text box and click Next 9. Configure the transport settings: For IP click the drop-down Session Protocol menu and select Tieline Codecs and ensure the Any check-box is not selected, then click Next.
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Merlin PLUS User Manual v1.4 Important Note: When Tieline Codecs is the Session Protocol selected (using Tieline session data), the Return Audio Port is automatically configured as UDP audio port 9010 by default for the second audio stream. Click to configure: ·...
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4. Enter the name of the connection in the text box, then click Next 5. Configure the transport settings: For IP click the drop-down Session Protocol menu and select Tieline Codecs and ensure the Any check-box is not selected, then click Next.
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8. Enter the name of the IFB audio stream connection in the text box and click Next 9. Configure the IFB transport settings: For IP click the drop-down Session Protocol menu and select Tieline Codecs and ensure the Any check-box is not selected, then click Next.
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Merlin PLUS User Manual v1.4 Important Note: When Tieline Codecs is the Session Protocol selected (using Tieline session data), the Return Audio Port is automatically configured as UDP audio port 9010 by default for the second audio stream. Click to configure: ·...
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Merlin PLUS User Manual v1.4 13.Configure the transport settings: For IP click the drop-down Session Protocol menu and select Tieline Codecs and ensure the Any check-box is not selected, then click Next. Important Note: When Tieline Codecs is the Session Protocol selected (using Tieline session data), the Return Audio Port is automatically configured as UDP audio port 9020 by default for the third audio stream.
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16.Enter the name of the second IFB audio stream connection in the text box and click Next 17.Configure the IFB transport settings: For IP click the drop-down Session Protocol menu and select Tieline Codecs and ensure the Any check-box is not selected, then click Next.
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1. By default, port 9000 is the port specified for the first IP audio stream connection made by any Tieline codec; port 9010 is used by default for a second IP connection. Merlin PLUS default ports for mono connections 3 to 6 are 9020, 9030, 9040 and 9050.
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8. Enter the name of the audio stream connection in the text box and click Next. 9. Configure the transport settings for this connection: For IP click the drop-down Session Protocol menu and select Tieline Codecs and ensure the Any check-box is not selected, then click Next.
· Failover and SmartStream PLUS redundant streaming are not available with SIP connections. · When connecting to a Tieline G3 codec using SIP you need to manually select the G3 audio reference level in the codec. To do this select SETTINGS >...
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· The default UDP audio port when using SIP for a peer-to-peer connection is 5004 in Tieline codecs. To contact a codec that is behind a firewall or NAT-enabled router, it is essential that this and all other relevant ports are open and forwarded to the other device.
SIP call and a default peer-to-peer program will be loaded. There is no Tieline session data transferred during SIP calls to assist with configuring the codec. Important Notes: · Remember to lock an answering program in a codec when answering multiple SIP calls.
2. Click Download Event Log and select a location to save the log file. Clearing Logs This option should only be used if instructed to by Tieline support staff. To clear all event and other logs in the codec via the front panel see the...
The codec supports both in-band and out-of-band data depending on the connection transport and algorithm you are using. RPTP data is automatically enabled when using the Tieline Music or MusicPLUS algorithms over any transport. Over IP it is also possible to enable synchronized out-of- band data using any algorithm.
3. Click Yes in the confirmation dialog. 22.26 Upgrading Codec Firmware To download the latest codec firmware visit http://www.tieline.com/Support/Latest-Firmware. Manual Firmware Upgrades The following procedure explains how to perform codec firmware upgrades with a downloaded firmware file saved to your PC.
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Merlin PLUS User Manual v1.4 Inputs Panel for Input Adjustments Important Note: Tieline codecs have different input configurations, therefore the image shown may not reflect the number of inputs displayed in your codec Web-GUI. Feature Description Settings button Click to adjust input Name, Type and IGC.
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Click to open the codec user manual in a new browser, or view support information (Note: the codec name displayed will vary by product type) 3 Support website link Click to visit the support page on the Tieline website. 4 Email Support Click to email Tieline support. 5 Event Logs...
An IP address is a unique address to identify a device on a TCP/IP network. Your codec uses dual IP protocol stacks to allow your codec to work on both IPv4 and IPv6 networks. Tieline codecs support both DHCP (default) IP addressing and static IP addresses for dialing IPv4 connection endpoints.
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ISDN Answer Configs are used to determine how codec ISDN modules will behave when answering ISDN calls. The following image explains the difference between answering calls from Tieline codecs sending session data, and non-Tieline codecs making sessionless ISDN calls. Codecs sending Tieline Session Data contain all the information required to connect, e.g.
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· Expected dialing behaviors, e.g. if B channels should bond or not, and whether audio streams need to use Dial and Answer Route tags. · The type of call being received by the codec, e.g. Tieline (with Tieline Session Data) versus non-Tieline sessionless calls.
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Dial Route and Answer Route tags allow you to associate a B channel (or channels) in a Config with a particular incoming audio stream from either Tieline or non-Tieline codecs. This is not necessary in simple point-to-point ISDN audio stream configurations, however it is very useful in multiple audio stream codecs using multiple B channels.
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Config 1-4 in ISDN Answer. You can also select the default algorithm. For example, if a call from a non-Tieline codec is being received via B Channel 1 on Module 1 (i.e. no Dial Route has been specified in the dialing codec): 1.
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If it fails to receive Tieline session data within 5 seconds (i.e. a non- Tieline codec is calling, or a Tieline codec with session data disabled), it will use the settings in the ISDN Answering Config instead.
Tieline session data and sessionless ISDN calls at different times. They can also support connections using other transports such as IP or POTS. The following example shows how a Tieline codec can be configured to answer up to 4 separate mono ISDN calls at different times from both Tieline and non-Tieline codecs, as well as two mono IP audio streams.
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The default Config settings for POTS modules are designed to suit Tieline codecs. These settings will need to be adjusted to connect to non-Tieline POTS codecs or connect in Analog Phone mode.
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Calls are answered based on the POTS Answer settings in Config 1 & 2. Adjustments to these Config settings are not normally necessary when connecting between Tieline codecs. They are usually adjusted when connecting to non-Tieline codecs over POTS (see...
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· Expected dialing behaviors and encoding, e.g. whether audio streams use Route tags and which algorithm is used. If you answer a call from a non-Tieline codec you will need to create an answering "Config" to determine which module in the codec will answer the call and the settings used when connecting.
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· POTS Answer Config settings are applied to POTS Codec connections and not Analog Phone connections. · When receiving a call from a Tieline codec with session data enabled (i.e. not Sessionless), the algorithm setting from the dialing codec overrides the setting in the POTS Answer Config menu.
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POTS connections. When dialing Tieline to Tieline over POTS using the Merlin or Genie family of codecs, you can configure a Dial Route in the dialing codec's program and a corresponding Answer Route in the answering codec's program. This will ensure a particular audio stream is routed between two codecs consistently.
Config 1 or 2 in POTS Answer. You can also select the default algorithm. For example, if a call from a non-Tieline codec is received via POTS Module 1 (i.e. no Dial Route has been specified in the dialing codec): 1.
SIP provides superior interoperability between different brands of codecs due to its standardized protocols for connecting devices and is intended to be used when connecting Tieline codecs to non- Tieline devices. Devices primarily use SIP to dial another device’s SIP address and find its location with a minimum of fuss.
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· Failover and SmartStream PLUS redundant streaming are not available with SIP connections. · When connecting to a Tieline G3 codec using SIP you need to manually select the G3 audio reference level in the codec. To do this select SETTINGS >...
1. Open the HTML5 Toolbox Web-GUI and click Settings in the Menu Bar, then click Help to display the Help panel. 2. Click Download Logs. 3. Save the file to your computer and then send it as a .zip file to Tieline support via support@tieline.com Download Event Logs Event logs can be downloaded from the codec and viewed in your browser.
Merlin PLUS User Manual v1.4 Clearing Logs This option should only be used if instructed to by Tieline support staff. To clear all event and other logs in the codec via the front panel, see the Reset and Restore Factory Default Settings...
The codec supports both in-band and out-of-band data depending on the connection transport and algorithm you are using. RPTP data is automatically enabled when using the Tieline Music or MusicPLUS algorithms over any transport. Over IP it is also possible to enable synchronized out-of- band data using any algorithm.
Merlin PLUS User Manual v1.4 Upgrading Codec Firmware 23.16 To download the latest codec firmware visit http://www.tieline.com/Support/Latest-Firmware. Firmware Upgrades The following procedure explains how to perform codec firmware upgrades with a downloaded firmware file saved to your PC. 1. Open the Toolbox HTML5 Web-GUI and click Settings in the Menu Bar, then click Firmware to display the Firmware panel.
Overview of Tieline Algorithms 1. The Tieline Music algorithm is optimized for audio bit rates as low as 19.2kbps with only a 20 millisecond encode delay. It offers 15 kHz mono from 24kbps to 48kbps. 2. Tieline MusicPLUS delivers up to 20 kHz mono from 48kbps upwards. It can also deliver up to 20 kHz stereo from 96kbps upwards, offering huge savings on your IP data bills and outstanding audio quality.
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For example, if MP2 algorithms are used, program delays will be much longer than when using Tieline Music or MusicPLUS algorithms. This is due to the additional inherent encoding delays involved when using MP2 algorithms. This can be a major consideration for live applications that integrate remotes into a broadcast.
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A fixed jitter buffer is preferable over satellite connections to ensure continuity of signals. CAUTION: If a Tieline codec connects to a device that is using non-compliant RTP streams then the last fixed setting programmed into the codec will be enabled (default is 500ms).
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FEC is being used. Fixing Jitter Buffer Settings The default jitter-buffer setting in Tieline codecs is 500 milliseconds. This is a very reliable setting that will work for just about all connections. However, this is quite a long delay and we recommend that when you set up an IP connection you test how low you can set the jitter-buffer in your codec.
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FEC according to the available uplink bandwidth at each end for best performance. As an example, if you want 15 kHz mono (using the Tieline Music Algorithm) you will need at least a 24kbps connection for audio. Adding 100% FEC will add another 24kbps making your bit rate 48kbps plus some overhead of around 10kbps is required.
2. Use the navigation buttons on the front panel to select Connect and press the button. 3. Select IP and press the button. 4. Select Tieline (or Sessionless) and press the button. 5. Select Peer-to-Peer and press the button. 6. Use the down navigation button to select Setup and press the button.
– ensuring critical session data (including dial, connect and hang-up data) will be received reliably. The default session and audio port settings in Tieline codecs, for both TCP and UDP connections, are outlined in the Installing the Codec at the Studio section of the manual.
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(port used by the local codec to receive audio from the remote codec). It is also possible to configure the send and return audio ports for a codec using Tieline session data to establish IP connections. This may be required because some firewalls require symmetric port configuration.
"Tieline Codec" Port Configuration If using the Tieline Codec setting for call establishment (i.e. Tieline session data is enabled), you can also change the default audio ports if required. · The default value for the Send (audio) Port is 9000 ·...
5 Clear Logs Deletes codec event and log history. Note: This should only be performed if instructed to by Tieline support staff. Important Note: After restoring factory defaults, always reboot the codec using the Reboot Codec function, not by removing power from the codec.
Tieline recommends that the racks in which codecs are installed are thoroughly evacuated to ensure proper airflow from the bottom to the top. Where space is available, a 1RU gap between codecs will assist in minimizing internal temperature build up.
The following instructions are intended to help you configure your internet connection and Tieline codecs at the studio to enable incoming calls over the internet from a remote Tieline codec. It is assumed that you have a basic understanding of your IP network and how to configure IP devices. If...
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The following table lists the firewall ports you need to open for each model of Tieline codec if they are dialing your router at the studio. If the remote codec is also connected to a LAN with a firewall you may also need to open the ports at the remote end of the link to connect successfully.
IP address and you know this address. If you dial the studio using a cell-phone data network at the remote site you will not normally experience any firewall or port blocking issues at the remote end of the link using default Tieline ports.
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Diagnosing Port Blocking via the Remote Codec LQ If you attach your Tieline codec at the remote site to a LAN with access to the internet you can often dial and connect to the studio without any problem. It is less likely that a firewall will block outgoing TCP and UDP ports.
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How do I determine which end is blocking data flow? Tieline test codec firewalls have the default Tieline TCP and UDP ports open. You can dial into these test codecs (or other codecs you know are configured correctly) from your recently configured studio and remote codecs and use the LQ readings to diagnose whether your studio or remote codec firewall is blocking your data packets.
(DHCP) and static. Most ISPs assign a dynamic public IP address by default, which can often change without you knowing. This is suitable for a quick demo of your Tieline codec, but for a permanent installation you will need to request a permanent static public IP address.
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Port Forwarding: Tieline TCP and UDP Port Settings For your Tieline Codec to communicate over the public internet an IP Address alone is not sufficient. In TCP/IP and UDP networks the codec port is the endpoint of your connection. Ports are doorways for IP devices to communicate with each other.
256 kbps for a studio codec and 64 kbps for a field unit connection. 5. Use good quality equipment to connect your codecs to the internet. (Tieline successfully uses Cisco® switching and routing equipment.): · If you are using a DSL or ADSL connection make sure you purchase a high quality modem that can easily meet your speed requirements.
Connecting Tieline ISDN to other Codecs To dial from a Tieline codec to a non-Tieline codec it is necessary to disable ‘Session Data’ and use an algorithm like G.722 or MPEG Layer 2 for compatibility. The same settings must be configured at both ends for: ·...
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7. For bonded "MPEG1-L2" connections select "CCS IMUX". 8. Complete the profile setup. The codec is now ready to dial or answer. Configuring the Tieline Codec to Dial the Equinox over ISDN 1. Press the HOME button to return to the Home screen and select Connect > ISDN.
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Dialing from the WorldCast Equinox Important Note: Configure ISDN Answer Config settings in the codec before attempting to dial from the Equinox to the Tieline codec. Select the following settings in the Tieline codec in one of the Configs (see...
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5. Press the right arrow on the "Enter" button and navigate to "Interface". Push the down arrow on the "Enter" button to select this menu. 6. Use the "Enter" button and navigate to the type of interface you are using. Note: During Tieline tests we used an "Internal TA".
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10.Use the "Enter" button and keypad to enter the "ID 1" and "ID 2" (Directory/MSN) numbers if required. 11.The codec should now be configured. Configuring the Tieline Codec to Connect to the CDQ Prima 1. Press the HOME button to return to the Home screen and select Connect > ISDN.
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32kHz sampling. Important Note: Configure ISDN Answer Config settings in the codec before attempting to dial from the Equinox to the Tieline codec. Select the following settings in the Tieline codec in one of the Configs (see ISDN Answering Configuration for more detail): ·...
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It has also been noted that the CDQ Prima codec will not connect if no audio is present when dialing. It may connect Prima > Tieline, but not Tieline > Prima. If audio is present, the codec should connect and stay connected even if audio is removed subsequently. The J-Stereo light on the Prima may also flash when in this mode.
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Dialing from the Mayah Sporty Important Note: Configure ISDN Answer Config settings in the Tieline codec before attempting to dial from the Equinox to the Tieline codec. Select the following settings in the Tieline codec in one of the Configs (see...
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Merlin PLUS User Manual v1.4 Configuring the Tieline Codec to Dial the Xstream over ISDN 1. Press the HOME button to return to the Home screen and select Connect > ISDN. 2. Navigate to Setup and press the button. 3. Select Session Type [Sessionless] > Select Dial Route [None] > Number of B Channels [Choose 1B or 2B] >...
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Dialing from the Zephyr Xstream Important Note: Configure ISDN Answer Config settings in the codec before attempting to dial from the Xstream to the Tieline codec. Select the following settings in the Tieline codec in one of the Configs (see...
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Merlin PLUS User Manual v1.4 25.7.5 Connecting to Comrex Matrix ISDN To connect your Tieline codec to a Comrex Matrix rack mount codec: 1. Use the G.722 algorithm. 2. Connect using only one 64Kbps ISDN B Channel (bonding of G.722 over two ISDN B channels is not possible).
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· DN/MSN > Enter the "SPID" and "DN" numbers if required in your region, e.g. a SPID is normally required in the US. 10. Navigate up to Apply Settings and press the button. Dialing from the Tieline Codec Program Dialing 1. If you have saved the ISDN program as previously instructed, press the HOME button to return to the Home screen and select Connect >...
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Dialing from the Comrex Matrix Important Note: Configure ISDN Answer Config settings in the codec before attempting to dial from the Comrex Matrix to the Tieline codec. Select the following settings in the Tieline codec in one of the Configs (see...
In addition you can use “Answer Routes” and 'site-specific' module settings in Genie Distribution and Merlin PLUS to route incoming calls to specific codec outputs. (Note: Merlin codecs can also be configured to accept 2 ISDN calls from non-Tieline codecs and would use similar settings).
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B channel then that’s all you need to do. If, however, you want each non-Tieline codec to use the same B channel and be routed to the same codec output consistently, you must configure this in the site config for the ISDN module via Settings >...
POTS Operation Precautions POTS performance is greatly affected by the quality of the line being used. Precautions must be taken to ensure the Tieline codec is not sharing the line with other devices. Please remove these possible sources of interference: ·...
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Telco. POTS Exchange Problems On most good POTS lines, Tieline codecs can achieve a 28.8kbps connection at a line quality of approximately 50% or greater. If you are not able to achieve this level of operation, you may have a problem with your line, or the line at the other end of the connection.
Simply place the ADSL/DSL filter between the POTS line and your codec to remove the interference. 3. Tieline USA has a POTS test codec you can dial on +1-317-913 6911 to facilitate line tests at each end of your connection to diagnose line problems.
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No user serviceable parts are contained in this product. If damage or malfunction occurs, contact TIELINE Pty Ltd for instructions on repair or return. This equipment cannot be used on a telephone company provided coin service. Connection to Party Line service is subject to state tariffs.
Web Browser Interface or from the Tieline website <http://tieline.com>. You may request a copy for the open source software on DVD by contacting our support team on +61 (0)8 9249 6688. Tieline Pty Ltd will charge a small handling fee for distribution of this software.
Sample Frequencies Sample 16kHz, 32kHz, 44.1kHz, 48kHz Frequencies Algorithms Tieline Music, Tieline MusicPLUS, G.711, G.722, MPEG-1 Layer 2, MP3, LC-AAC, HE-AAC and HE-AACv2, AAC-LD, AAC-ELD, Opus, 16/24 bit aptX Enhanced IP (uncompressed) Linear PCM16/24 bit 48kHz sampling Data and Control Interfaces USB 2.0 Host port on the front panel...
Connector Pins Important Notes: · The codec cannot send RS232 data to, or activate relays on Tieline G3 codecs. · It is important that you enable serial port flow control within the codec. Flow control regulates the flow of data through the serial port. If disabled, data will flow unregulated and some may be lost.
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