[asterisk-users] No audio after a reinvite changing codec

[Matteo Campana]

Hi all,
we have a problem with a reinvite sent by our SIP provider to change audio
codec due to the recognition of a fax tone.
After that the SIP call session has been established (INVITE and 200 OK) we
have the following codec situation:

UAC                                        ASTERISK UAS | ASTERISK UAC
                         PROVIDER
g711      <---------------------->           g711      |       g729
<--------------------------->  g729
                        rtp
                                rtp

After a while, we have the reinvite sent by the SIP provider with g711 in
the SDP.
So asterisk need to change audio codec from g729 to g711 and correctly we
see on debug the following line:
"Oooh, we need to change our audio formats since our peer supports only
g729" and asterisk send back 200 OK to the provider.
At this point we have one way rtp audio:

UAC                                        ASTERISK UAS | ASTERISK UAC
                         PROVIDER
g711      ---------------------->           g711      |       g711
--------------------------->  g711
                        rtp
                                rtp

So the problem is that UAC does not hear audio at all.
Any idea?

(Asterisk version: 1.4.33.1).

Thanks in advance,
Matteo
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