UCM630xA Series User Manual
Version 1.0.9.10
Table 39: SIP Extension Configuration ParametersMedia
Use NAT when the UCM630xA is on a public IP communicating with devices hidden
behind NAT (e.g., broadband router). If there is one-way audio issue, usually it is
related to NAT configuration or Firewall's support of SIP and RTP ports. The default
setting is enabled.
By default, the UCM630xA will route the media steams from SIP endpoints through
itself. If enabled, the PBX will attempt to negotiate with the endpoints to route the
media stream directly. It is not always possible for the UCM630xA to negotiate
endpoint-to-endpoint media routing. The default setting is "No".
Select DTMF mode for the user to send DTMF. The default setting is "RFC4733". If
"Info" is selected, SIP INFO message will be used. If "Inband" is selected, a-law or u-
law are required. When "Auto" is selected, RFC4733 will be used if offered, otherwise
"Inband" will be used.
If the phone has an assigned PSTN telephone number, this field should be set to
“User=Phone”. “User=Phone” parameter will be attached to the Request-Line and
“TO” header in the SIP request to indicate the E.164 number. If set to “Enable”, “Tel”
will be used instead of “SIP” in the SIP request.
When present in an INVITE request, the alert-Info header field specifies and
alternative ring tone to the UAS.
Enable or disable T.38 UDPTL support.
Enable SRTP for the call. The default setting is disabled.
Select jitter buffer method.
Disable: Jitter buffer will not be used.
Fixed: Jitter buffer with a fixed size (equal to the value of "jitter buffer size")
Adaptive: Jitter buffer with an adaptive size (no more than the value of "max
jitter buffer").
NetEQ: Dynamic jitter buffer via NetEQ.
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