No audio after a reinvite changing codec

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Asterisk Users 4 Comments

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

4 thoughts on - No audio after a reinvite changing codec

  • We experience the same thing. The solution we use is to not change codecs in the middle of a call. I assumed it was an issue with our upstream.

  • Inviato da iPhone

    Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling ha scritto:

    Hi Eric,
    this behavior is an asterisk bug or asterisk can never change the codec “on the fly”?

    Thanks,
    Matteo

  • On Mon, Jun 20, 2011 at 11:58 PM, Matteo Campana
    wrote:

    Hi,
    I’m out of the office this week, next Monday I will send the debug to the
    list.
    However I think It’s strange asterisk behavior: it says 200 OK after a
    re-invite by the provider, but stops to send rtp.

    Regards,

    Matteo