Asterisk 13 - Opus transcoding

Asterisk Asterisk Support
1

Hi,
I’m rejecting calls due to codec mismatch No joint capabilities for ‘audio’ media stream between our configuration((g722|g729)) and incoming SDP((opus)).
Does Asterisk support opus transcoding in Asterisk 13.17.2 or even in later release?

Thanks

2

That transcoding will require two steps, with signed linear as an intermediate. It is most likely that you are “negotiating” G.729 but don’t have the g729 codec or enough licences.

3

The codec translation paths show intermediate steps are involved.

opus:48000 To g729:8000 : (opus@48000)->(slin@48000)->(slin@8000)->(g729@8000)
opus:48000 To g722:16000 : (opus@48000)->(slin@48000)->(slin@16000)->(g722@16000)

4

Thinking about it, I think you are actually telling Asterisk to negotiate a codec that the peer doesn’t support. Although SIP allows asymmetric codecs, Asterisk will only use codecs that both it and the peer support. The transcoding happens between the two peers, not between the peer and Asterisk.