[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:comment-tabpanel&focusedCommentId=245884#comment-245884 ] 

David Kuehling commented on ASTERISK-28230:
-------------------------------------------

Can now confirm that the issue is also present with Asterisk 16.1.1.  I need the following workaround to allow GXV3140 to display video (tested using the echo test):

{code:none}
Index: asterisk-16.1.1/res/res_rtp_asterisk.c
===================================================================
--- asterisk-16.1.1.orig/res/res_rtp_asterisk.c
+++ asterisk-16.1.1/res/res_rtp_asterisk.c
@@ -4603,6 +4603,7 @@ static int rtp_raw_write(struct ast_rtp_
 
 		/* If the abs-send-time extension has been negotiated determine how much space we need */
 		abs_send_time_id = ast_rtp_instance_extmap_get_id(instance, AST_RTP_EXTENSION_ABS_SEND_TIME);
+		abs_send_time_id = -1;
 		if (abs_send_time_id != -1) {
 			/* 4 bytes for the shared information, 1 byte for identifier, 3 bytes for abs-send-time */
{code}



> 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
>
> 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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