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Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling <[email protected]> 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: [email protected] >> [mailto:[email protected]] 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 >> <[email protected]> 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 -- _____________________________________________________________________ -- 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
