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

[Matteo Campana]

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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