JDS Uniphase SmartClass User Manual page 104

Chapter 11 VoIP Testing
Specifying test settings
Setting
Called Type (H.323)
Outbound Alias Type
(SIP)
SIP Vendor (SIP)
Device Type (SCCP)
Device Name Type
(SCCP)
Device Name
(SCCP)
Endpoint ID Type
(MGCP)
Endpoint ID (MGCP)
3 Press the right arrow to move to the Audio/Codec settings, and then specify the settings.
The following table describes the settings.
Setting
Audio codec
Frame Interval
Jitter buffer
Transmit Source
Silence suppression
RTP Diffserv Value
RTP Port Minimum
RTP Port Maximum
The next step depends on the CC Standard selected.
– For H.323, go to
– For SIP, go to
– For SCCP or MGCP, go to
86
Description
If required, this sets the Called Party Type information element in the
H.323 setup message for outgoing calls: Unknown, International, National,
Network Specific, Subscriber, Abbreviated.
Select the method for outbound dialing: Dial by Phone Number or Dial by
Name/URL/Email.
Specify the vendor: Standard SIP, Nortel SIP, Huawei SIP.
Select the device type.
Select one of the following: Automatic based on MAC address, or User
Defined
If Device Name Type is set to "User Defined", enter a device name.
Select one of the following: user defined or automatic based on the IP
address (for example,
aaln/01@[10.50.20.2]).
If Endpoint ID Type is set to "User Defined", enter the ID of the endpoint.
Description
This selects which codec will be used.
Set the speech per frame. This is the number of milliseconds of speech per
transmission frame when using a sample based codec (such as G.711).
Set the jitter buffer size. This is the number of milliseconds of speech that
will be collected before an attempt will be made to play the speech back.
This allows lost, late, or out-of-sequence packets time to arrive and be reas-
sembled before playback.
Select the source of transmission: Voice conversation (transmits and
receives live voice), IP voice announce (the unit repeats a sequence of
words including the calling party's IP address), Tone
Enable or disable.
Enter a value to indicate the Voice IP Quality of Service Differentiated Ser-
vices (DiffServ) code point.
The value will occupy a 6-bit field in the packet headers of RTP stream voice
packets and will indicate how packets are treated at each hop. You can
specify a number from 0 to 63 to indicate the per-hop behavior.
Specify the RTP port minimum number.
The real-time transport protocol (RTP) port number allows you to identify
voice traffic versus other traffic. Some systems only accept RTP traffic on
certain port numbers. (It must be a factor of 2.)
Specify the RTP port maximum number. (It must be a factor of 2.)
step 4.
step 5.
step 6.
SmartClass TPS User's Guide
22035456, Rev 001 June 2014