[asterisk-bugs] [JIRA] (ASTERISK-27702) allow video/h264 breaks symmetric rtp

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27702?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=242388#comment-242388 ] 

Asterisk Team commented on ASTERISK-27702:
------------------------------------------

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].

> allow video/h264 breaks symmetric rtp
> -------------------------------------
>
>                 Key: ASTERISK-27702
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27702
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip
>    Affects Versions: 14.5.0, 15.2.2
>         Environment: linux x64
> Verified on 
> asterisk 15.2.2 with pjsip 2.7.2
> asterisk 14.5 and pjsip 2.6
>            Reporter: Jørgen H
>              Labels: pjsip
>
> Adding codec h264 in allow-list result in that rtp_symmetric gets ignored and asterisk sends rtp auto to the ip-address in the sdp even when h264 is not in use.
> Config:
> rewrite_contact = yes
> rtp_symmetric = yes
> force_rport = yes
> direct_media = no
> allow = !all,opus,g722,alaw,ulaw,h264
> SDP from answering telephone when audio works without h264 in allow list:
> v=0
> o=MxSIP 0 1 IN IP4 10.24.5.115
> s=SIP Call
> c=IN IP4 10.24.5.115
> t=0 0
> m=audio 3000 RTP/AVP 9 8 0 101
> a=rtpmap:9 G722/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=silenceSupp:off - - - -
> a=fmtp:101 0-15
> a=ptime:20
> a=sendrecv
> SDP when h264 is enabled and audio does not work (audio is sent to 10.24.5.115):
> v=0
> o=MxSIP 0 1 IN IP4 10.24.5.115
> s=SIP Call
> c=IN IP4 10.24.5.115
> t=0 0
> m=audio 3000 RTP/AVP 9 8 0 101
> a=rtpmap:9 G722/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=silenceSupp:off - - - -
> a=fmtp:101 0-15
> a=ptime:20
> a=sendrecv
> m=video 0 RTP/AVP 99



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