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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