On Fri, 4 Sep 2009, Tim Panton wrote: > On 4 Sep 2009, at 07:53, Armin Schindler wrote: > >> 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 >> > > I've got g722 running through 1.4.22.2 with the patch set that targets 1.4.7 > > Calls from our java iax softphone come in as IAX2 in g722 and leave via SIP > to a g722 conference service. > seems to work ok. No transcoding, recording etc, and the jitterbuffer is > _off_ since it's a VoIP to VoIP call.
Yes, when is disable jitterbuffer (I had even forcejitterbuffer=yes in iax.conf and jbforce=yes in sip.conf), it works here too. I don't know how the jitterbuffer is doing it, but could it be possible that the RTP info of g722 (stated 8000 but it actually 16000) is confusing the jitterbuffer? 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
