[OpenSIPS-Users] Input on my loadbalancer configuration

geoffreymina at gmail.com geoffreymina at gmail.com
Thu Nov 13 13:05:53 CET 2008


Bogdan (and list),
Again, thank you very much for taking the time to answer my questions. It  
is people like you that enable the open source community to thrive and  
grow. I really appreciate your efforts.

I have gone through and made your suggested changes and things seem to be  
working quite well for what I am trying to accomplish.

My last and final question to the group (for now ;)... how do you determine  
the dimensions of an OpenSIPS deployment? I am going to be running two of  
these systems with automatic failover. They will be running on Dell 1950  
with 8 cores of CPU (2.0Ghz) and 8G or RAM on CentOS x_86/64. OpenSIPS is  
the only extra process running on the systems. Will this configuration  
handle 1,000 concurrent calls, 10,000 concurrent calls, etc? Will my limits  
be in Call Setups Per Second? How does one go about estimating the capacity  
of OpenSIPS?



Thanks,
Geoff

On Nov 13, 2008 5:02am, Bogdan-Andrei Iancu <bogdan at voice-system.ro> wrote:
> Hi Geoff,
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> geoffreymina at gmail.com wrote:
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> Thank you very much for taking the time to look over my configuration. I  
just want to make sure of something. I replied to my own original message  
with a greatly enhanced configuration. I realized the first was missing a  
huge amount of logic after studying up on OpenSIPS for 2 days. Were you  
commenting on the original message, or the second message? Based on my  
testing, i experienced slightly different results than you described.
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> My comments were generally speaking (for a LB topic) and not strictly in  
regards to a particular script.
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> What I am seeing (based on the second config) is that only the initial  
INVITE falls into the route(1) block, which is the way I intended it. This  
means only the INVITE requests are routed via the ds_select_dst() call to  
the dispatcher. All subsequent messages fall into my loose_route() check  
and are simply relayed via t_relay().
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> yes, because you still so record_route() - if you remove this, you will  
process only the initial INVITEs - the ACK, BYEs will not even pass thorugh  
your server.
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> I included a little snippet of the logging I do via the xlog calls. One  
thing that confuses me is that based on my configuration and my logs, I  
never explicitly relay the TRYING or OK messages. I set up my  
onreply_route[1], but all I do is log that I got the reply. I did this  
because regardless of what I do here, the UAC which requested the INVITE  
gets the TRYING and OK messages properly. Is there something in the tm.so  
that implicitly handles these, or am I missing some big picture element  
here.
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> replies are automatically routed back on the reverted path of the request  
- you do not need to route them explicitly.
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> Regards,
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> Bogdan
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> Thanks!
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> Geoff
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> ################ BEGIN LOG SNIPPET ########################
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> New request - M=INVITE RURI=sip:+15552021000 at 10.2.14.100 F=sip:[REMOVED]  
T=sip:+15552021000 at 10.2.14.100 IP=10.2.252.190  
ID=5fe66b3e04cbdd991217a4426afa42f8@[REMOVED]
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> Recording Route info
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> Method is an INVITE, fetching next from dispatcher
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> Reply - S=100 D=Trying F=sip:[REMOVED] T=sip:+15552021000 at 10.2.14.100  
IP=10.2.252.181 ID=5fe66b3e04cbdd991217a4426afa42f8@[REMOVED]
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> Reply - S=200 D=OK F=sip:[REMOVED] T=sip:+15552021000 at 10.2.14.100  
IP=10.2.252.181 ID=5fe66b3e04cbdd991217a4426afa42f8@[REMOVED]
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> New request - M=ACK RURI=sip:+15552021000 at 10.2.252.181 F=sip:[REMOVED]  
T=sip:+15552021000 at 10.2.14.100 IP=10.2.252.190  
ID=5fe66b3e04cbdd991217a4426afa42f8@[REMOVED]
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> Recording Route info
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> Loose route has returned true, attempting routing.
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> Setting up reply handler and relaying request
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> New request - M=BYE RURI=sip:+15552021000 at 10.2.252.181  
F=sip:6789050671 at connectfirst.com T=sip:+15552021000 at 10.2.14.100  
IP=10.2.252.190 ID=5fe66b3e04cbdd991217a4426afa42f8 at connectfirst.com
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> Recording Route info
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> Loose route has returned true, attempting routing.
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> Setting up reply handler and relaying request
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> Reply - S=200 D=OK F=sip:[REMOVED] T=sip:+15552021000 at 10.2.14.100  
IP=10.2.252.181 ID=5fe66b3e04cbdd991217a4426afa42f8@[REMOVED]
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