[asterisk-dev] RTCP work going on :: Project Pinefrog

[Olle E. Johansson]

Friends,
Just a head up that I've got sponsorships from a few companies in the community to start working on improving our RTCP support in chan_sip. This work will hopefully be done during december. The current RTCP reports in 1.4 are kind of random. There are improvements in trunk, but there's still room for additional work.

I want to add support for RTCP SDES and BYE. When we get a BYE, we should produce a final report that summarizes the "call". I also want to try to aggregate some stats on peer level, so we can see stats for a particular "trunk" or device in regards of packet loss and jitter for the last <configurable> amount of calls.

The RTCP SDES name is a tag for the RTP session, since the SSRC might change. This will help us to separate RTCP when we reinvite and transfer calls. One "call" might have multiple SDES sessions and multiple SSRCs. I haven't got a clear plan on how to handle this. If an outbound call has a first media session with one device, then suddenly gets reinvited to another device, we need to see this as two media sessions with different RTCP stats. Well, let's start with being able to separate them and see what happens next.

Funny enough, I realized that this projects code name is "pinefrog". Who knew?

Any ideas, feedback or contributions are as always welcome.

/O