[SR-Users] Asterisk and Kamailio Integration CAPACITY ISSUE

Olle E. Johansson oej at edvina.net
Mon Sep 16 09:42:21 CEST 2013

16 sep 2013 kl. 08:41 skrev Hassan Wajahat <hassanw89 at gmail.com>:

> Hi,
> I followed your tutorial and the setup is running seamlessly. But the question is that I read somewhere that asterisk can make up to 300 to 400 concurrent sip calls. Higher than that will cause problems. Will that issue appear here as well since you are using asterisk in the sip calling process??

That's not correct. I have servers in production with Asterisk 1.4 running up to 1500 simultaneous calls to IVRs. I have successfully tested over 10.000 simulatenous channels on a single server - after that the Gigabit Ethernet was filled up and I had plenty of CPU left.

If you use conferencing or transcoding, CPU usage per call goes up. We had an issue with more than 700 meetme's but that was a limitation in the DAHDI infrastructure that Digium could not solve, so we had to switch to app_konference.

No magic, old versions of Asterisk, a few year old tests so I think if we have 10 gig Ethernet and modern servers we can really tests the limit of Asterisk. That would be fun. 

I suspect that the limitation we'll see in chan_sip is the shared thread that reads all messages from the wire. chan_sip is not multithreaded in the part that reads and parses messages from the network. Remember that every call in Asterisk is executed in a private thread so we do handle multiple cores nicely - it's just this part of chan_sip that I think we'll see hit the roof when it reaches 100% usage of a single core. I haven't been able to force that to happen yet though.


More information about the sr-users mailing list