On Thu, 3 Sep 2009, Tilghman Lesher wrote: > On Thursday 03 September 2009 02:47:05 Armin Schindler wrote: >> Hello, >> >> I try to move our asterisk installation (3 Asterisk servers in different >> offices connected using IAX and a lot of SIP phones, as well as ISDN >> connections using CAPI) to use G.722 instead of G.711. >> >> Asterisk 1.4.25.1 is used with the G.722 patch (the fixed one, which solves >> the gain problem). >> So SIP-to-SIP and to ISDN there is no problem. G.722 itself works and >> transconding to G.711 for ISDN also works good. >> But when I make a connection through IAX to another asterisk (having >> allow=g722 to really use G.722 in IAX) the voice is 'broken'. >> >> I also work on G.722 for twinklephone and encountered a special thing about >> G.722: It has a sample rate of 16000, but it announced as 8000 in SDP. >> Since I have similar problem with my G.722-twinkle implementation, it looks >> like the RTP and/or jitterbuffer code has a problem with that. >> Did I miss something here or is this really a bug? > > You missed that the IETF has a typo in the specification, stating that G.722 > is to be stated as 8000, even though it's 16000. This will remain, due to > backwards compatibility concerns. Please see RFC 3551, section 4.5.2. > http://www.apps.ietf.org/rfc/rfc3551.html#sec-4.5.2
No, I didn't miss that. See my text. I mentioned this because I think this might be the reason of the problem and the incorrect handling in jitterbuffer, if it is the jitterbuffer. It is just a guess, since everything else seems to work good. The question is why does G.722 via IAX has problems. Is anyone using it and can say it works in his setup? Armin _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
