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)

The reason that I store  QoS data in realtime, is that the CDR is usually gone or frozen at the time that we freeze the RTP channels and get the last QoS data. The QoS reports can’t thus be included in CDR, you have to merge it in automatically later in your database.There’s still a lot to do, but please test it so I get some sort of feedback.For testing, don’t forget to run the “rtcp debug” cli command so you can see what’sgoing on in the RTCP channel.


  • Yes, this work will be ported to trunk and hopefully merged soon.
  • No, we don’t support RTCP XR or MOS in this work
  • No, I have no reason or funding to adapt it to 1.6.x at this point.
  • No, the RTPAUDIOQOS channel variable is not changed. You will get more data than before – for many more calls.

This work is funded to 20% by companies in the community. If you want to cover the80% that’s still not funded, please contact me by e-mail: oej@edvina.net.