> -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Russell Bryant > Sent: Tuesday, July 24, 2007 11:06 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] SIP jitter buffer and asterisk native bridge > > Damon Estep wrote: > > Anyone know the answer? Has it been validated with packet captures, or > > code review? > > All of the timing information should be passed across the bridge in all of > the > frames that come in over RTP. I can't say I verified this with packet > captures, > but I did look for this in the code review for the jitterbuffer code in > 1.4. I > know there is explicit code to ensure this is the case.
Russell, Thanks a bunch. So in theory the media gateway at the far end should be able to properly jitter buffer the entire RTP path from the ATA via asterisk, correct? Would this be the same in 1.2 and it 1.4? The best practice in the example given would be to rely on adaptive jitter buffers at the ATA and the media gateway, and not force jitter buffers in the SIP<>SIP asterisk bridge (1.4) Damon _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
