[asterisk-users] asterisk 1.2 and UDP packet numbering on bridged channels (for jitter buffering)?

[Damon Estep]

http://www.asterisk.org/node/48317 does a nice job of explaining the 1.4
jitter buffer, however it raised a question in my mind.

 

In 1.2 (and also 1.4), when asterisk bridges 2 SIP channels, are the UDP
RTP packets renumbered on transmit, or is the original sequence number
preserved in the UDP header?

 

A comment is made on the referenced blog that jitter buffering is best
implemented at the endpoint, where additional jitter will not be added
by another IP link. This is logical thinking, but only possible if the
bridging function in Asterisk preserves the source call leg UDP packet
numbering in the terminating call LEG UDP RTP packet stream.

 

If the effect of the Asterisk SIP to SIP bridge is such that the UDP
headers are re-created on transmit it is likely that the packet
sequencing is the order in which Asterisk transmitted the packets, which
is may not be the order in which the original source UA transmitted them
due to jitter in the IP link on the first half of the bridged call.

 

Can anyone provide an authoritative answer on how asterisk sequences UDP
RTP packets on the transmit leg of a bridged SIP call (known based on
actual testing or code review)? 

 

Or maybe there is information I lack that makes this a silly question,
such as where the SIP RTP sequence number is stored in the packet (ie:
not in the UDP header?) :-)

 

Thanks!

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