[asterisk-dev] Please test Pinefrog-rtcp-1.8

[Olle E. Johansson]

Hi!

I've spent a few days porting my RTCP work for Asterisk 1.4 to Asterisk 1.8, which has a different RTP subsystem. When this works and we're satisfied with these changes, it will be much more simple to port it to 11 and trunk than starting with 1.4.

I haven't changed the actual data in RTCP - number of packets, jitter, loss, latency - but I've changed the way we handle RTCP and exchange messages as well as added extended reporting. Asterisk 1.8 already had manager events, but I've added Call Quality Records.

Earlier discussions about CDRs on this mailing list concluded that there's no way possible to store this data for every call scenario in the CDRs. That's why I decided to create CQRs. The Asterisk SIP channel now stores records for every RTP stream in a database, with pointers to both SIP call IDs and Asterisk channel IDs (uniquieid and channel name). Optionally we can log them to the "cqr" logging channel.

Many of these changes is done in the RTP core engine and the Asterisk RTP subsystem, making it available for other channels that use RTP too.


A few GUI developers seems to have taken the challenge to do something useful with this data, including Edgar Landivar in the Elastix project. On the list of my ideas:

- Monitoring of SIP trunks. What's the quality of the latest 10 calls, the latest 100? Is it getting worse or better? Mark the sip trunk useless in worst case and fail over to another one in the dialplan.
- Monitoring of long calls in call centers - use the manager reports and monitor call quality. If an agent fills the ethernet cable with file transfers and cool youtube videos, don't send any new calls
- Checking call quality for individual peers, reporting issues
- Nagios/Icinga integration


I'm not fully done, still looking for some issues. The branch is fully useful and ready for testing. If you want to help testing and give feedback the time is now. Read the README.pinefrog and send me feedback.

Thanks to NORDICOM Norway AS that has funded this work.

Instructions
=========
svn checkout http://svn.digium.com/svn/asterisk/team/oej/pinefrog-rtcp-1.8

configure, build and install like any other Asterisk branch!

I'm looking forward to your feedback!

Best regards,
/O