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Olle has an RTCP branch for 1.4 ready to test

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The awesome branch Olle has been working on to fix RTCP in Asterisk is ready for testing - well worth a look.

His full article is available here:

test rtcp test branch based asterisk 1.4

I’ve created a test branch for patches hidden in several Asterisk development branches - all based on Asterisk 1.4

* RTCP improvements from pinefrog-1.4
* “Sip show chanstats” cli command
* The branch pinequality-* giving you the manager “sipchannel” event to check QoS

This branch is now open for testing and I need feedback. Among the improvements you’ll find:

* Manager QoS events during a call and after a call
* Improved RTCP - it now works for p2p bridge in RTP, which means that we will get RTCP stats for many, many more sip calls
* RTCP over NAT improvements - if Asterisk is behind NAT, we will now kick-start RTCP from the remote end by sending a first “emtpy” RTCP packet to open a NAT port.
* QoS reports to realtime storage after each call - one report per call leg (The amount of data and the names will change)


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