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...erent 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 i...

...rnal, small jitter buffer and it drops samples from the audio stream when there is high jitter in the network. The bandwidth is cheap now so for me the only reason to use transcoding is where I have low-bandwidth-high-jitter links. Sangoma said they will not fix it and we had to go back to software transconding. Do you have any experience with using Digium cards in such scenario? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131122/45b53c7a/attachment.html>