asterisk users - Jul 2007 - SIP jitter buffer and asterisk native bridge

There is a theory that says that jitter buffers should not be used until
the end of the voice path where jitter might be introduced. With that in
mind, and in this scenario, the jitter buffers should reside at the ATA
and media gateway;

 

ATA (SIP UA)  <> ASTERISK NATIVE BRIDGE <> MEDIA GATEWAY (SIP TO
TDM)

 

That raises a question about the Asterisk Native Bridge; Are the UDP RTP
packets bridged in such a way that out of order packet arrivals between
the ATA and asterisk can still be buffered and corrected at the media
gateway, or are the RTP sequence numbers re-written by the Asterisk
native bridge so the media gateway is now unaware that they are not in
the same order as they were initially transmitted?

 

Anyone know the answer? Has it been validated with packet captures, or
code review?

 

Thanks a bunch!

 

Damon

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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 Bryant
Software Engineer
Digium, Inc.