Andrew Kohlsmith [EMAIL PROTECTED] wrote: > [EMAIL PROTECTED] wrote: > > My qualification is having worked on the IAX2 jitter buffer, > > consequently having studied how audio flows from the received frames > > through the jitter buffer and then via ast_translate() into the codec. > > > Hmm... having worked on the IAX2 jitter buffer, can you tell us why > trunking and jitter buffers don't get along? When trunking with nufone I > get ... interesting... audio if I have a jitter buffer enabled. :-) > Getting back to loss concealment for a moment, it seems to me that we could do something like the following:
* Every 20ms, call a scheduled function that inserts a "silent" voice frame into the stream. The frame would be marked as "bogus" in some way and would be timestamped appropriately. * The jitter buffer should then remove the "duplicate" voice frames, leaving a constant 20ms stream of either voice data or silence. * The individual codecs should then either spot the frame's "bogus" marker and deal with it as a dropped frame or, if the codec can't do reconstruction, process the frame as silent audio. I expect that a silent frame would sound much the same as a dropped frame (with no reconstruction) anyway. Does that sound feasible with the current framework? My initial inspection of the SIP/IAX2 code says that it should be, although it'd introduce a fair amount of overhead. -- _/ _/ _/_/_/_/ _/ _/ _/_/_/ _/ _/ _/_/_/ _/_/ _/ _/ _/ _/_/ _/ K e v i n W a l s h _/ _/ _/ _/ _/ _/ _/ _/_/ [EMAIL PROTECTED] _/ _/ _/_/_/_/ _/ _/_/_/ _/ _/ _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users