[asterisk-dev] Please test my RTCP work!

Wolfgang Pichler wpichler at yosd.at
Thu Jan 21 04:24:55 CST 2010


Hi,

i would be happy to test it - and maybe to integrate the new manager events
with asterisk-java - to be able to create some fancy live quality monitor...

Testing does only make sense on system with some load - so the question is -
how stable is it already ? Is it ready for testing on a production system ?

And - how long would it take for you to port it to 1.6.x ?

best regards,
Wolfgang

2010/1/21 Olle E. Johansson <oej at edvina.net>

> Friends,
> I've just created a combined test branch for work on RTCP that I would
> appreciate
> your feedback on.
>
> Thanks!
>
> /Olle
>
> http://svn.asterisk.org/svn/asterisk/team/oej/pinefrog-deluxe-rtcp-test/
>
> Test branch for patches hidden in several branches - all based on Asterisk
> 1.4
>
> ----------------------------------------------------------------------------------------------------------
>
> - RTCP improvements from pinefrog-1.4
> - "Sip show chanstats" cli command
> - pinequality-* giving you the manager "sipchannel" event to check QoS
>
> This branch is now open for testing and I need feedback. Among the
> improvements
>
> - Manager QoS events during a call and after a call
> - Improved RTCP - rtcp now works for p2p bridge in RTP, which means that we
> will get
>  RTCP 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's
> going on in the RTCP channel.
>
> FAQ
> ---
> Yes, this work will be ported to trunk and hopefully merged soon.
> No, I have no reason or funding to adapt it to 1.6.x at this point.
> No, the RTPAUDIOQOS is not changed at all. YOu might get more data now
> though.
>
> -------
> This work is funded to 20% by companies in the community. If you want to
> cover the
> 80% that's still not funded, please contact me off-list (oej at edvina.net).
>
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