[Asterisk-Dev] RTCP-'support'

[Filip Olsson]

Hello boys and girls,

I've posted a patch to
http://bugs.digium.com/bug_view_page.php?bug_id=0002863 that add's some
RTCP-support to rtp.c. It adds the ability to send/receive (and almost
understand) RTCP sender/receiver reports.

It's kind of hard to test if the receiver actually understands what we
send them. The way I made sure it was correctly understood by the remote
end was to make sure that such things as LSR and DLSR in RTCP RRs was
filled in correctly to reflect our SR. The RTP-stack in 42Networks'
DRG-series understand them and responds correctly, according to
ethereal.

The whole point of this patch was to dig out some statistics on the
quality(delay/jitter/loss) of the RTP-streams and then dump it to
somewhere useful. I've written a small patch to chan_sip so that some
averages/maxs/mins are saved in the userfield of the CDR. John Todd has
put forth a much better solution that is more generic(not just RTP)
http://lists.digium.com/pipermail/asterisk-dev/2004-May/004180.html.
But atleast it's a start to have the stuff in the CDR. 
It would be even more useful to(on a per call basis) be able to log a
RTP-stream to a separate file that describes the whole stream/call in
more detail so one can monitor the stats as the call goes on(for people
that like fancy graphing).

RTCP and RTP is described in RFC3550, it's the best documentation I've
found.

Anyways, please test the patch and give me some feedback. I'm in
#asterisk/-dev/-bugs under my very secret nick folsson.

Don't even think about patching your production systems with this one,
it'll most certainly blow up your machine and get you fired.

//Filip Olsson