Matteo Campana
2011-Jun-13 16:55 UTC
[asterisk-users] No audio after a reinvite changing codec
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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Matteo Campana
2011-Jun-15 12:15 UTC
[asterisk-users] No audio after a reinvite changing codec
HI list, no idea?? :) 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-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110615/2864080d/attachment.htm>
Larry Moore
2011-Jun-16 14:31 UTC
[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 <mailto: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 > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110616/ded4611d/attachment-0001.htm>
Eric Wieling
2011-Jun-16 14:37 UTC
[asterisk-users] 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.> -----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 > > > > > -- > > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by > http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory > webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >
Matteo Campana
2011-Jun-17 21:36 UTC
[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 >> >> >> >> >> -- >> >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by >> http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory >> webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Larry Moore
2011-Jun-18 04:40 UTC
[asterisk-users] No audio after a reinvite changing codec
On 18/06/2011 5:36 AM, Matteo Campana wrote:> > 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 >The problem reported occurs after a fax tone is detected, one might expect T.38 or G711 to be used to handle the fax, not G729! There is no configuration file information for each of the nodes/peers, no debug of each peer involved nor a trace of the packets hence no one will have ideas! Larry.