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? Thanks, 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
