On Tuesday 03 August 2004 07:18, [EMAIL PROTECTED] wrote: > I am the source for that statement. Is that a problem? ;-)
Not at all. :-) But I do thank you for taking the time to write a few paragraphs explaining what's going on in the current code. It's certainly something I didn't know before and it really takes out an advantage of using iLBC over something like GSM -- the former has lower bandwidth requirements but the increased processor overhead and lag might be causing another problem I've been seeing. I've switched back to GSM for now to see if the audio quality issues goes away. > 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. :-) "Hey there, how are you today" turns into "He....ythe....rehow....areyou....tod....ay" > In Asterisk, bridging is "self-clocked" by the incoming frames. So unless > a frame arrives, nothing happens. Each arriving frame gets pushed through > the codec decode function. But if a frame doesn't turn up... well, > then... nothing will happen. Aside from easier implementation is there any advantage to having the audio streams self-clocked? -A. _______________________________________________ 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
