baresip and webrtc sip clients specify opus codec like this:

a=rtpmap:96 opus/48000/2.
a=fmtp:96 stereo=1;sprop-stereo=1.

whereas sems does it like this:

a=rtpmap:105 opus/48000.

i don't know what the specs say, but looks like this incompatibility
prevents sems using opus with the above mentioned sip clients.

-- juha
_______________________________________________
Semsdev mailing list
[email protected]
http://lists.iptel.org/mailman/listinfo/semsdev

Reply via email to