No path to translate from PJSIP/webrtc_client_1

Asterisk Asterisk WebRTC
1

I have configured websocket and working fine according to document but when placing call from webrtc to pjsip extension then after picking call immediately it disconnect and in asterisk console Im getting this log

[Feb  5 00:13:58] WARNING[3596]: channel.c:5765 set_format: Unable to find a codec translation path: (opus) -> (ulaw)
[Feb  5 00:13:58] WARNING[3596]: channel.c:5765 set_format: Unable to find a codec translation path: (ulaw) -> (opus)
[Feb  5 00:13:58] WARNING[4408][C-0000001e]: channel.c:6756 ast_channel_make_compatible_helper: No path to translate from PJSIP/webrtc_client_1-0000003a to PJSIP/6001-0000003b

Thank you

2

It means what it says. It can’t transcode between opus and ulaw, probably because you don’t have the codec_opus module. You can select the Sangoma provided binary version using “make menuselect” when building Asterisk, or you can opt not to use opus.

1 Like
3

Thank you it worked for me

4

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