Asterisk might be unable to transcode rtp type from downstream to upstream, or vice versa. There's a bug reported here, for asterisk 12 or above, using chan_sip. https://issues.asterisk.org/jira/browse/ASTERISK-25676 It says that you could avoid the bug by using chan_pjsip, but you still encounter it? Turn `core set debug 5` to see whether you have `Unsupported payload type received` like I once did? rgds,
On Wed, Mar 15, 2017 at 1:40 AM Faheem Muhammad <faheem2...@gmail.com> wrote: > Hi, > I'm facing strange issue while establishing inbound calls from SIP trunks. > Provider A is sending (G729,Alaw,uLaw) offer and asterisk dial the peer > with its preferred codec order(G729,aLaw, uLaw). The peer's phone send the > codec list as (uLaw, speex) in 200 OK replay. The Peer's phone has selected > only uLaw and speed in this case. > > Ideally Asterisk should establish the call on uLaw codec, but Asterisk > establish the call with two codec for this call. For downstream RTP is > established with G729 and for upstream RTP is established with uLaw codec. > This behavior cause the one way audio for some phones like Eyebeam 1.5.9 > but Phonerlite latest version allow it and there is no audio issue. > > Is it normal SIP RFC 3261 behavior or there is something wrong with codec > negotiation or transcoding? > > I'm using Asterisk 13.14.0 with realtime chan_pjsip compiled with bundled > pjproject on centos 6.8_x64. I have tested it with Asterisk 11.x with > chan_sip and it works fine. > > Please advise me how can I setup the call based on late negotiation > mechanism? > > Thank you! > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users