[asterisk-bugs] [JIRA] (ASTERISK-29258) chan_sip: Audio stream rejected, Other stream present: Invalid SDP.

[Alexander Traud (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-29258?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Alexander Traud updated ASTERISK-29258:
---------------------------------------

    Attachment: invite_video.xml

> chan_sip: Audio stream rejected, Other stream present: Invalid SDP.
> -------------------------------------------------------------------
>
>                 Key: ASTERISK-29258
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29258
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/CodecHandling
>    Affects Versions: 13.38.1, 16.16.0, 18.2.0
>            Reporter: Alexander Traud
>            Assignee: Alexander Traud
>              Labels: patch
>         Attachments: invite_video.xml, video_with_rejected_audio.patch
>
>
> When Asterisk receives a call with audio and video, but the audio stream is rejected because of incompatible formats. This can happen when the callee is misconfigured (not even G.711 as audio codec). Then, Asterisk continues with the video stream only. So far, so good. However, Asterisk answers with invalid SDP:{code}m=audio 49170 RTP/AVP 
> m=video 49172 RTP/AVP 34{code}This is a violation of [RFC 3264 section 6|https://tools.ietf.org/html/rfc3264#section-6]
> bq. To reject an offered stream, the port number in the corresponding stream in the answer MUST be set to zero.
> and [RFC 4566bis|https://tools.ietf.org/html/draft-ietf-mmusic-rfc4566bis] because at least one format must be specified.
> The cause is the re-architecture of the media-format handling introduced with Asterisk 13 in commit [a2c912e|https://github.com/asterisk/asterisk/commit/a2c912e9972c91973ea66902d217746133f96026], six years ago. The issue got worse with the fix for ASTERISK-24543 (commit [d343a25|https://github.com/asterisk/asterisk/commit/d343a25173e846fe2474700ed1849df4d74d88f3] and [3f72015|https://github.com/asterisk/asterisk/commit/3f720155b7c8b3089c59cbfc78150aff4efc8240]) because that fix was incomplete.
> The attached patch makes sure, {{p->caps}} are only asked for audio formats when they are going to be appended. In other words, the append of {{p->caps}} is guarded with the same condition now. The attached XML file is for the app SIPp. Because I do not get the Asterisk Test Suite running, I cannot create a complete test case.



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