Taken from previous threads: "
You can’t. WebRTC and normal phones use different incompatible SDP, thus why there are options to configure this. Noone has working on trying to make it work for both.
You cannot use the same endpoint for a phone and webrtc*
I have defined two endpoints in pjsip, one for a softphone phone and the other for webrtc.
I can make calls from webrtc to a softphone but not vice-versa.
Is call initiation from both endpoints in all ways possible?
If configured correctly, yes. You haven’t provided the configuration, or stated what actually happens when it doesn’t work, or provided logging so noone can help any further with that aspect.
Thank you for your swift response, i get the following error:
pp_dial.c:3381 dial_exec_full: Had to drop call because I couldn’t make PJSIP/100-00000004 compatible with PJSIP/200-00000005
200 - softphone
100 - webrtc phone
Below are my configurations
local_net = 0.0.0.0/24
; All other transport parameters are ignored for wss transports.
That would mean each side is using different codecs, and there is no translation path between them. Likely one side is opus and the other isn’t, and you don’t have a codec_opus to transcode between the two. You could confirm this by removing opus from the allow list on both, or selecting codec_opus in menuselect when installing Asterisk.
I removed opus and its now now working, thank you
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