[asterisk-biz] Asterisk 1.4 RTCP support - want to fund fixing it?

[Olle E. Johansson]

Friends in the Asterisk community.

Recent testing has proved to me that the Asterisk 1.4 RTCP support, the protocol that measures media quality factors such as round trip time, jitter and packet loss, is not up to standard. There are many bug reports in the tracker about bad calculations, but the fact is that we're basing all RTCP stats on random data today, if anything at all.

The RTCP packet sent from the remote end contains several information blocks. Our code assumes the first one is of one type and the next one is of another type, which is not correct. Thus it mixes sender and receiver reports. In trunk, RTCP maintains aggregated data about the packet loss and jitter for a call, something 1.4 does not do, it seems to report the last reported value for round trip and jitter. Asterisk should also parse the RTCP SDES information block and send one, as that block contains the overall session identifier for a call, the one that we should use to aggregate data at the end of the call to present the RTPAUDIOQOS etc variables.

I am searching funding from Asterisk service providers, users and integrators to fix this. I want to fix 1.4 so it shows correct data per call and verify that trunk works correctly. 1.4 is in use today in many production environments and trunk is, hopefully, going to be the next long-term support Asterisk version (pending decision). Unless someone that provides funds wants me to check 1.6.x versions and provide fixes for them, I won't focus on those releases.

Please contact me OFF LIST if you are interested in funding this work. If a few interested parties can fund one work-day each, so I can spend at least three full days on this issue, I think a lot can be done to improve our RTCP support.

Thanks,
/Olle


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* Olle E. Johansson - oej at edvina.net
* Asterisk Training http://edvina.net/training/