[asterisk-users] SIP jitter buffer and asterisk native bridge

[Damon Estep]

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] 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