[asterisk-bugs] [JIRA] (ASTERISK-28047) SDP has extra info resulting in call issues

[Will (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-28047?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Will updated ASTERISK-28047:
----------------------------

    Description: 
@jcolp sent me here to post this from the dslreports forums.

When placing a call from a cisco phone registered to a Cisco Unified Communications Manager (11.5) across a sip trunk to an asterisk server out a gvsip trunk, the call fails after a reinvite contains SDP with extraneous information. The endpoint is setup to allow nothing but g711ulaw, yet without explicitly max_video_streams=0 on the endpoint, SDP contains the m=video and offers h264. This causes the call to drop after ~30seconds.

  was:
Joshua Colp sent me here to post this from the dslreports forums.

When placing a call from a cisco phone registered to a Cisco Unified Communications Manager (11.5) across a sip trunk to an asterisk server out a gvsip trunk, the call fails after a reinvite contains SDP with extraneous information. The endpoint is setup to allow nothing but g711ulaw, yet without explicitly max_video_streams=0 on the endpoint, SDP contains the m=video and offers h264. This causes the call to drop after ~30seconds.


> SDP has extra info resulting in call issues
> -------------------------------------------
>
>                 Key: ASTERISK-28047
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28047
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 15.6.0
>         Environment: debian 9.5, NAF's gvsip fork with oauth
>            Reporter: Will
>            Severity: Minor
>         Attachments: extensions.conf, gvsip trace1.txt, gvsip trace2 with codecs disabled.txt, gvsip trace3 max_video_streams=0.txt, pjsip.conf, rtp.conf
>
>
> @jcolp sent me here to post this from the dslreports forums.
> When placing a call from a cisco phone registered to a Cisco Unified Communications Manager (11.5) across a sip trunk to an asterisk server out a gvsip trunk, the call fails after a reinvite contains SDP with extraneous information. The endpoint is setup to allow nothing but g711ulaw, yet without explicitly max_video_streams=0 on the endpoint, SDP contains the m=video and offers h264. This causes the call to drop after ~30seconds.



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