Video Calling Asterisk 16.15

I successfully implemented video calling on Asterisk 13.32 and FreePBX14. Used the same devices on a different system running FreePBX15 and Asterisk 16.15 and no video is present. Exact same asterisk sip settings for video. Am I missing something in the new asterisk version? Not sure if it is significant but I do see an error when looking in the logfiles

WARNING[10005][C-00000005] chan_sip.c: Don’t know how to indicate condition 36

I do see the call using RTP Audio and Video.

1879	[2022-08-10 11:56:13] VERBOSE[15449] netsock2.c: Using SIP RTP Audio TOS bits 184	
1880	[2022-08-10 11:56:13] VERBOSE[15449] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.	
1881	[2022-08-10 11:56:13] VERBOSE[15449] netsock2.c: Using SIP RTP Audio CoS mark 5	
1882	[2022-08-10 11:56:13] VERBOSE[15449] netsock2.c: Using SIP RTP Video TOS bits 136	
1883	[2022-08-10 11:56:13] VERBOSE[15449] netsock2.c: Using SIP RTP Video TOS bits 136 in TCLASS field.	
1884	[2022-08-10 11:56:13] VERBOSE[15449] netsock2.c: Using SIP RTP Video CoS mark 4	
1885	[2022-08-10 11:56:13] VERBOSE[10005][C-00000005] app_dial.c: PJSIP/2201-00000003 is ringing	
1886	[2022-08-10 11:56:13] VERBOSE[10005][C-00000005] app_dial.c: PJSIP/2201-00000003 is ringing	
1887	[2022-08-10 11:56:15] VERBOSE[10005][C-00000005] app_dial.c: PJSIP/2201-00000003 answered SIP/2220-00000004	
1888	[2022-08-10 11:56:15] VERBOSE[10006][C-00000005] bridge_channel.c: Channel PJSIP/2201-00000003 joined 'simple_bridge' basic-bridge <5ab7557c-9789-4374-984c-08fba0b03880>	
1889	[2022-08-10 11:56:15] VERBOSE[10005][C-00000005] bridge_channel.c: Channel SIP/2220-00000004 joined 'simple_bridge' basic-bridge <5ab7557c-9789-4374-984c-08fba0b03880>	
1890	[2022-08-10 11:56:15] VERBOSE[12417] netsock2.c: Using SIP RTP Video TOS bits 136	
1891	[2022-08-10 11:56:15] VERBOSE[12417] netsock2.c: Using SIP RTP Video TOS bits 136 in TCLASS field.	
1892	[2022-08-10 11:56:15] VERBOSE[12417] netsock2.c: Using SIP RTP Video CoS mark 4	
1893	[2022-08-10 11:56:15] WARNING[10005][C-00000005] chan_sip.c: Don't know how to indicate condition 36

Condition 36 is stream topology changed. There has presumably been a re-invite. That is probably on the channel that isn’t reporting this warning, so might be the PJSIP one, in which case the “pjsip set logger on” output, from the full log, will be helpful. In case it is the chan_sip one, or there are two chan_sip channels involved, “sip set debug on” output may be useful.

I’d suggest that, when doing anything complex, it is much better if both sides use the same technology, as that will allow native bridging.

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Actually, thinking about it, I think the ability to add streams mid-call is only supported in chan_pjsip, in which case both side will have to use that driver for it to work, and that probably explains the warning.

The SDP trace might explain why it is happening if you didn’t intend it, but if you did intend it, I don’t think you can use chan_sip.

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Good idea. I’ll convert the SIP to pjsip and test. Thank you!

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