[asterisk-users] PJSIP Codec Negotiation Issue

[Benoît Panizzon]

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?

Also reading:
https://wiki.asterisk.org/wiki/display/AST/PJSIP+Advanced+Codec+Negotiation

does not explain what I see.

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-Benoît Panizzon- @ HomeOffice und normal erreichbar
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