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

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26932?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=236456#comment-236456 ] 

Asterisk Team commented on ASTERISK-26932:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> [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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