[asterisk-bugs] [JIRA] (ASTERISK-28230) res_rtp_asterisk: abs-send-time extension added with Asterisk 15.5.0 breaks GXV3140 video telephony

[Joshua C. Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-28230?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua C. Colp updated ASTERISK-28230:
--------------------------------------

    Summary: res_rtp_asterisk: abs-send-time extension added with Asterisk 15.5.0 breaks GXV3140 video telephony  (was: abs-send-time extension added with Asterisk 15.5.0 breaks GXV3140 video telephony)

> res_rtp_asterisk: abs-send-time extension added with Asterisk 15.5.0 breaks GXV3140 video telephony
> ---------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-28230
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28230
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>    Affects Versions: 15.5.0, 15.7.1, 16.1.1
>         Environment: Debian Linux Stretch amd64; Asterisk compiled from sources.
>            Reporter: David Kuehling
>            Assignee: Unassigned
>              Labels: pjsip, webrtc
>         Attachments: asterisk-15.4.1-conversation.txt, asterisk-16.1.1-conversation.txt
>
>
> Hi,
> I noticed that upgrading from Asterisk 15.4.1 to 15.7.1 my Grandstream GXV3140 stopped displaying h264 video stream.  It just shows a black picture, shows correct video data rate, but displays video frame rate as 0 FPS.
> Bisecting the issue I found that the failure happens after commit eb52c70f5b143a96228ffe91c5215b76f48ec355, which implements ASTERISK-27831.
> This is also corroborated by some source changes I tested to work around the issue introduced by the above commit:
> If I overwrite the value of max_send_time_id, forcing it to -1 in rtp_raw_write(), then video works again.
> I don't really understand what's going on here.  The commit eb52c70f5b143a962 looks like it would support some negotiation to enable max-send-time support, however there is no indication that such a negotiation is happening.  Certainly my GXV310 does not claim any support in its SDP payload.  Also I did not find any way to disable the feature using the config files, nor any way to get log output about any negotiation taking place.
> This is the SDP payload that is transmitted by the GXV3140:
> {code}
> v=0
> o=10 8001 8000 IN IP4 172.20.20.16
> s=SIP Call
> c=IN IP4 172.20.20.16
> t=0 0
> m=audio 31514 RTP/AVP 118 9 0 8 101
> a=sendrecv
> a=rtpmap:118 L16/16000
> a=ptime:10
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> m=video 31516 RTP/AVP 99
> b=AS:320
> a=sendrecv
> a=rtpmap:99 H264/90000
> a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4Ecg==; max-br=320
> {code}



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