Performance measurement

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Measure the performance of rtmplite for simple (1) video recording, (2) 
playback streaming, (3) audio only, (4) siprtmp translation for audio call, (5) 
real-time two-party call.

Original issue reported on code.google.com by voiprese...@gmail.com on 3 Feb 2011 at 7:07

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Original comment by voiprese...@gmail.com on 3 Feb 2011 at 7:09

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Also reported by tom hensel <tom@interpol8.net>on Jun 2, 2010.

hi,

i'm working as a technical consultant for a social network in germany.

we are currently looking for a sip/rtmp bridging solution to provide
telephony to the network.

i'm kindly asking you on experience in terms of stability and
scalability in a large scale deployment of rtmplite:
we are estimating 10k calls a day.

any info is much appreciated.

best regards,

Original comment by voiprese...@gmail.com on 4 Feb 2011 at 11:24

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Also reported by C.Savinovich <c.savinovich@itntelecom.com> on Jul 12, 2010.

Hello Sir, your program is an excellent idea, congratulations and many
thanks on your project.

I am a developer with good knowledge of sip/asterisk but very little flash.
I will appreciate your help in the following question, there are some things
I don't understand how I can make it work.  For what I gather in your
document, I see how I can create one demo, but I don't know if your gateway
supports many sip-2-rtpm calls at the same time, since this is intended to
work on a web site with maybe 20 calls at the same time.  Can you please
confirm me if this is possible?, also, would you be available for some
support if we need to?, thank you

Chris Savinovich

Original comment by voiprese...@gmail.com on 4 Feb 2011 at 11:48

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First, i really have 2 say that i am in love with this project. and if you guys 
need an asterisk consultant (6+ years) with some flex knowlodge for this, 
please let me know and i will happy 2 help.

tested in production. And sorry for my tarzan english :D

i manage to get ~20/30 concurrent agents SIP2RTMP - ASTERISK, using an asterisk 
based predictive dialer. After that the performance of the asterisk is really 
bad, the transcoding kill the asterisk pretty much.. I am surprise of the 
stability..

We need it more agents connected around 60~ so we went back 2 x-lite (shame).

agents using app-queue connected all the time using transcoding in the asterisk

i am using right now flash 11B with ULAW and rtmp_gevents. not for the phones 
yet haven't have the oportunity to bring it to production yet. but for our 
supervisors/monitors is working using app_queue and app_chanspy. the change in 
performance are good enough to make feel i can give it a shot at 60 agents soon 
(60 concurrent calls). 

Original comment by ncrip...@gmail.com on 31 Jul 2011 at 4:17

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Hi ncrip,

you are very welcome to contribute and the project
with your knowledge. your experiments are interesting, yes gevent
is definitely the lib to reduce CPU, but of course it needs some improvments

Original comment by in...@live-school.net on 31 Jul 2011 at 5:28

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Thanks a lot people... we will be happy to take any patches/contributes to the 
project... any suggestion for improving the performance are also welcome...

Original comment by kun...@twilio.com on 3 Aug 2011 at 5:59

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Some measurement results were reported in 
http://p2p-sip.blogspot.com/2011/04/performance-of-siprtmp-multitask-vs.html 
and gevent version improves the performance.

Original comment by kundan10 on 27 Oct 2011 at 3:13

• Changed state: Done