[asterisk-users] PJSIP Codec Negotiation Issue

[Joshua C. Colp]

On Fri, May 20, 2022 at 6:48 AM Benoît Panizzon <benoit.panizzon at imp.ch>
wrote:

> Hi List
>
> I have come over a codec negotiation issue.
>
> A (asterisk) is sending in INVITE containing
> * opus (type 107)
> * g722
> * alaw (type 8)
>
> B answers with 183 containing SDP
> * alaw
> a=sendrecv
>
> B then answer the call with 200 and NO SDP
>
> I suppose that result in B telling us, it only support alaw.
>
> But 'set rtp debug on' show B sending type 8 and A sending type 107.
> As the remote only announced to be capable of 8, shouldn't asterisk
> send type 8? Or even send a Re-Invite to tell it switches to alaw?
>

What is the specific issue that is happening? If it's that one call leg
negotiated at opus and the other at alaw, that is currently the way things
still work. Each call leg is still ultimately negotiated independently so
the A leg can be opus, and the B leg can be alaw. I hope that we're able to
eventually return to codec negotiation work to improve that with the
foundation put into place previously, but I don't know when that will
happen.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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