[asterisk-bugs] [JIRA] (ASTERISK-23327) chan_sip - RTP Packetization set in general section not applied when Dialing direct to a SIP URI

[Rusty Newton (JIRA)]

Rusty Newton created ASTERISK-23327:
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             Summary: chan_sip - RTP Packetization set in general section not applied when Dialing direct to a SIP URI
                 Key: ASTERISK-23327
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23327
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Channels/chan_sip/CodecHandling, Codecs/General, Core/CodecInterface
    Affects Versions: 1.8.11.1, 11.7.0
         Environment: CentOS release 6.5 (Final) x64
MySQL Database
Dell R320 Server and Dell R210II Server
            Reporter: Jarrod Sears
            Assignee: Jarrod Sears
            Severity: Critical


[Edit by Rusty]
Issue is easily reproduced by Dialing or originating calls to any SIP URI, where RTP packetization is specified for a codec allowed in the general section of sip.conf. The codec specified will be used, but the packetization will not be applied. It may happen with peers as well, but I was unable to reproduce it there.
[End edit]

Incoming calls to the server correctly negotiate to g729:60.

The sip.conf is set to:
disallow=all
allow=g729:60

The outbound leg of the server then sends a SIP invite out specifying g729:20.

I've tried setting up specific SIP peers and also using the general codec settings, both experience the same issue.

I have also tried using the SET(SIP_CODEC=G729:60) command, which does not work.

I have tried using the a server with Digium's g729 codec. I've also tried on a server without a g729 codec. Both experience the same issue.

I originally experienced the issue on the 1.8.11.1 (Asterisk 1.8.11-cert1) version that we are using generally for production. I recently installed a new copy of Asterisk 11.7.0 and have the same problem.




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