[OpenSIPS-Users] How to bill SIP session time correctly?

[Andrew Pogrebennyk]

On 29.07.2010 17:49, Olle E. Johansson wrote:
> And yes, I have tested the 2.000 channels in a lot of different settings, with various boxes and various pieces of test equipment. The 10.000 channels test was between two HP servers and we reached a limit on the GB ethernet interface, not the CPU.

2.000 channels with RTP agrees with my experience too (and I was 
monitoring RTP quality on all channels). BTW I came across this link: 
http://www.thirdlane.com/forum/10000-channels-on-asterisk-milestone-reached 
- were those 10.000 channels running the p2p RTP bridge so asterisk did 
not proxy the RTP stream after re-INVITE between endpoints?
I'm just slightly confused by this sentence: "SIP to SIP calls, the p2p 
RTP bridge, basically running a media proxy".

-- 
Sincerely,
Andrew Pogrebennyk