[asterisk-bugs] [JIRA] (ASTERISK-26932) [patch] SIP/SDP: No rtpmap for static RTP payload IDs

[Alexander Traud (JIRA)]

Alexander Traud created ASTERISK-26932:
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             Summary: [patch] SIP/SDP: No rtpmap for static RTP payload IDs
                 Key: ASTERISK-26932
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26932
             Project: Asterisk
          Issue Type: Improvement
      Security Level: None
          Components: Resources/res_pjsip_sdp_rtp
            Reporter: Alexander Traud
            Severity: Minor


If a lot of audio and video codecs are enabled (allowed), an outgoing call might hit the MTU bearer of 1300 bytes for SIP over UDP. In [SDP|http://tools.ietf.org/html/rfc4566#section-5.14], specifying the rtpmap for a [static RTP payload ID|http://www.iana.org/assignments/rtp-parameters] is a SHOULD but optional. This saves around 20 bytes per format. This affects G.722, G.711, G.729, and GSM for example. Consequently, more than 100 bytes can be saved even on typical deployments = around 10%.

FreeSWITCH does this on default but reports some broken clients implementations. Therefore, FreeSWITCH included the variable [verbose_sdp|http://wiki.freeswitch.org/wiki/Variable_verbose_sdp] to disable this feature. Consequently, although in different specifications, I coupled this new feature with [SIP Compact Form|http://tools.ietf.org/html/rfc3261#section-7.3.3]. Compact Form is disabled on default. When someone enables those compact headers, he is about reducing the message size and does benefit from this change here as well. This reduces the complexity of the attached patch to two Boolean statements.

This is a port of ASTERISK-25578 from from the SIP channel driver chan_sip to res_pjsip.



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