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

[Eric Wieling]

I don't know.

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
> Matteo Campana
> Sent: Friday, June 17, 2011 5:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] No audio after a reinvite changing codec
>
>
>
> Inviato da iPhone
>
> Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling
> <EWieling at nyigc.com> ha scritto:
>
> >
> > 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.
>
>
> Hi Eric,
> this behavior  is an asterisk bug or asterisk can never
> change the codec "on the fly"?
>
>
> Thanks,
> Matteo
>
>
>
>
> >
> >> -----Original Message-----
> >> From: asterisk-users-bounces at lists.digium.com
> >> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf
> Of Larry
> >> Moore
> >> Sent: Thursday, June 16, 2011 10:32 AM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [asterisk-users] No audio after a reinvite changing
> >> codec
> >>
> >> On 15/06/2011 8:15 PM, Matteo Campana wrote:
> >>
> >>      HI list,
> >>      no idea?? :)
> >>
> >>
> >>
> >> There not much substance in the information provided for an
> >> assessment to be made.
> >>
> >> I would suggest you capture the network traffic between UAC
> >> (g711) & Asterisk UAS ensuring the snap length is large enough to
> >> capture the whole packet and do the same with traffic between
> >> Asterisk UAC & Provider then use Wireshark and its
> telephony feature
> >> to analyse VoIP calls, check the flows, you may discover
> the problem
> >> this way!
> >>
> >> Larry.
> >>
> >>
> >>
> >>      M.
> >>
> >>
> >>      On Mon, Jun 13, 2011 at 6:55 PM, Matteo Campana
> >> <matteo.campana at gmail.com> wrote:
> >>
> >>
> >>              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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> >>
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