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 Post subject: Can number of concurrent calls be increased with openser?
PostPosted: Mon Jan 12, 2009 5:31 am 
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Joined: Wed Feb 21, 2007 11:52 pm
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Wondering if openser a signalling server and Asterisk a media/gateway increase the number of concurrent calls on the Asterisk box.

What good is openser beside the signalling?
(an example, lets say one openser machine and one asterisk box used as a media server/gateway).
Can it increase the number of concurrent calls that the media server can normally handle?
Assuming 250-300 calls as norm.

Need clarification please.


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PostPosted: Mon Jan 12, 2009 5:46 am 
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Joined: Thu Jun 22, 2006 2:19 pm
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Location: Devon, UK
You could use it to distribute calls in round-robin fashion to multiple instances of Asterisk (in VServer/OpenVZ containers, or real servers), thereby increasing your peak call load. You can do this fairly effectively with Asterisk alone however.

Some use Kamailio (as OpenSER is now known) for the slightly better NAT-traversal techniques it provides over Asterisk's capabilities, although I think Asterisk v1.6 has closed the gap. In these days of ISPs aggressively filtering and shaping traffic, establishing a VPN-tunnel is often the most reliable solution.

Another use for Kamailio is to relieve Asterisk of the processing load of registrations. See middletn's recent posts here for more info on that topic.


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PostPosted: Mon Jan 12, 2009 6:02 am 
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Thanks stavros, yeah that i agree when it comes to handling more stress of UACs registrations on a network. I think Asterisk is capable of handling all of that in a reasonable manner. Thanks for the clarification.
I implemented kamailio 1.4.3 with Asterisk and still couldn't figure out really what kamailio is doing besides just registration and load balancing if decide to throw in more servers.


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PostPosted: Mon Jan 12, 2009 11:42 pm 
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Location: Bovey, Devon, UK
Fronting asterisk with a SER allows you to take advantage of the superiour NAT capabilities. In addition, I've heard rumours that asterisk falls over with a lot of registrations

regards


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 Post subject:
PostPosted: Tue Jan 13, 2009 12:20 am 
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Joined: Mon Apr 30, 2007 6:43 am
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Location: Canada
If you manage to avoid processing the RTP stream and successfully reinviting most of the calls, you can handle over 1000 registrations and channels with asterisk. I am talking about almost 100% pure sip calls. It is possible if talk to your providers and work with them in order to do this.

Cheers


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