Hi Ovidiu and Bogdan,<div><br></div><div>I see your point. Thanks for clearing my thoughts about this issue!</div><div><br></div><div>Regards,</div><div>Ronald<br><br><div class="gmail_quote">On Sun, Feb 27, 2011 at 5:53 PM, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@opensips.org">bogdan@opensips.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Hi Ovidiu,<br>
<br>
I got your point - I was making a comment to complete your statement.<br>
<br>
I agree that rate limit and concurrent call limit are 2 protection mechanism that normally are more suitable in different scenarios.<br>
<br>
But more or less, this is up to the admin to decide :) (which to use and in what scenarios).<div><div></div><div class="h5"><br>
<br>
Regards,<br>
Bogdan<br>
<br>
Ovidiu Sas wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hello Bogdan,<br>
<br>
Just for the record, I am not against having more flexibility in<br>
building scripts and new features (like dynamic creation of pipes) are<br>
more then welcome. In my previous replies I was just pointing out<br>
that ratelimit-ing basic accounts and small trunks doesn't add more<br>
protection for the server itself or better subscriber management.<br>
Setting a ratelimit for a SIP account that has a channel limit of two<br>
is an overkill, IMHO.<br>
Ratelimit-ing really makes sense for large SIP trunks and whole SIP<br>
traffic (server protection).<br>
<br>
<br>
Regards,<br>
Ovidiu Sas<br>
<br>
On Fri, Feb 25, 2011 at 11:59 AM, Bogdan-Andrei Iancu<br>
<<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>> wrote:<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi Ovidiu,<br>
<br>
actually we were flirting for some time with extending the ratelimit module<br>
to allow dynamic creation of pipes - this will give you more flexibility in<br>
scripting and scenarios (like dynamic number of trunke, DB provisioning,<br>
etc).....But this was a bit postponed as there are other more important<br>
things to do and resources are limited ;)<br>
<br>
Regards,<br>
Bogdan<br>
<br>
Ovidiu Sas wrote:<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
There are two different things:<br>
a. channel limitation or concurrent call limit;<br>
b. ratelimit or cps limitation (cps = cals per second).<br>
<br>
With the dialog module, you limit _only_ the number of concurrent<br>
calls (a). How fast will a SIP trunk be saturated is up to the cps.<br>
If you have a limit of 30 calls and the calls are coming in at a rate<br>
of 1 cps, in 30s you will reach your limit. If the calls are coming<br>
in at a rate of 15 cps, you will reach the limit in 2s.<br>
<br>
With the ratelimit module, you limit _only_ the number of calls per<br>
second that are processed.<br>
If you set a cps limit of 5 cps and your incoming traffic is 100cps,<br>
then every second the first 5 calls will go through and the next 95<br>
will be rejected (based on TAILDROP algorithm). If the traffic is<br>
steady for 10s, then you will have 10x5=50 active calls and 10x95=950<br>
rejected calls due to cps limitation.<br>
<br>
Now, if you combine both (30 channels max and 5cps limit for incoming<br>
traffic at 100 cps), during the first 6s you will saturate the trunk<br>
(by accepting the first 5 calls and rejecting the other 95 every<br>
second) and all calls after that will be rejected.<br>
<br>
Hope this helps.<br>
<br>
<br>
Regards,<br>
Ovidiu Sas<br>
On Wed, Feb 23, 2011 at 12:51 PM, Ronald Cepres <<a href="mailto:rbcepres@gmail.com" target="_blank">rbcepres@gmail.com</a>><br>
wrote:<br>
<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
On Wed, Feb 23, 2011 at 6:10 AM, Ovidiu Sas <<a href="mailto:osas@voipembedded.com" target="_blank">osas@voipembedded.com</a>><br>
wrote:<br>
<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
If a virtual PRI is set up (23 channels for NA or 30 channels for<br>
Europe), again the cps doesn't really count. As soon as the virtual<br>
PRI is maxed out (in terms of channels) all subsequent calls will be<br>
rejected (and the cps will be 0).<br>
<br>
<br>
<br>
</blockquote>
Hi Ovidiu,<br>
Does that mean that if i have a concurrent call limit of 30 for a user<br>
and<br>
he bursts 500 cps, he still wouldn't exceed 30 cps?<br>
By the way, I use a method similar to the one posted in the tutorials to<br>
limit concurrent calls<br>
(link: <a href="http://www.opensips.org/Resources/DocsTutConcurrentCalls" target="_blank">http://www.opensips.org/Resources/DocsTutConcurrentCalls</a>)<br>
Thanks!<br>
Regards,<br>
Ronald<br>
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<br>
</blockquote>
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</blockquote>
--<br>
Bogdan-Andrei Iancu<br>
OpenSIPS eBootcamp - 28th February 2011<br>
OpenSIPS solutions and "know-how"<br>
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<br>
<br></div></div>
-- <br><div><div></div><div class="h5">
Bogdan-Andrei Iancu<br>
OpenSIPS eBootcamp - 28th February 2011<br>
OpenSIPS solutions and "know-how"<br>
<br>
<br>
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