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

[David Kuehling (JIRA)]

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

David Kuehling updated ASTERISK-28230:
--------------------------------------

    Attachment: asterisk-16.1.1-conversation.txt
                asterisk-15.4.1-conversation.txt

SIP conversation corresponding to phone call to echo test number with video enabled.  Only SIP port was captured (no RTP).

Captured once with asterisk 15.4.1 (the last version without abs-send-time) and once with asterisk 16.1.1.

> 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: David Kuehling
>              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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