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Hi, Although this maybe considered out of scope here, but I'll ask anyway. Opus has remarkable flexibility for changing encoder bitrate during a call. Are there any suggestions about how bandwidth/capacity between the two endpoints can be measured/estimated during a call so that the outgoing bitrate can be adjusted accordingly? Thanks, Manpreet. -------------- next part -------------- An

...FAILED. DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time... UNSUPPORTED OR FAILED. DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01... UNSUPPORTED OR FAILED. DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid... UNSUPPORTED OR FAILED. DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. DEB...

I need to get hooked up with this class, I could have students doing projects for homework :) Interested in RTCP? j On 6/26/23 7:45 PM, TTT wrote: > > I’m in training, so I have to demonstrate something SIP related.  I > figure it would be cool to hack a call, hanging it up while in > progress from outside Asterisk.  Doing so will demonstrate > use/knowledge of ARI, AMI, SIP,

...10 10:45:24] a=mid:video [May 10 10:45:24] a=extmap:2 urn:ietf:params:rtp-hdrext:toffset [May 10 10:45:24] a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time [May 10 10:45:24] a=extmap:4 urn:3gpp:video-orientation [May 10 10:45:24] a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 [May 10 10:45:24] a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay [May 10 10:45:24] a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type [May 10 10:45:24] a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/video-tim...