<div dir="ltr">I believe bandwidth shuld be checked in media-processor since it's aware of number of ports used and codecs in each of them.<br></div><div class="gmail_extra"><br><div class="gmail_quote">2015-10-28 17:40 GMT+03:00 Guillaume <span dir="ltr"><<a href="mailto:tetram100@hotmail.fr" target="_blank">tetram100@hotmail.fr</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div dir="ltr">Thanks for your return and your advices. <br><br>In fact the bandwidth criteria would have meaning only if kamailio is a media gateway (otherwise we could only have a rough estimation with the negotiated codec as you said) and in this case getting the bandwidth from the interface will make sense.<br><br>I was also thinking about using a limit for the number of call per user but it would be redundant with the dialog module, doesn't it?<br><br>At the end do you think it's a good idea to add such features to kamailio?<br><br><br>Guillaume<br><br><br><div><hr>To: <a href="mailto:sr-dev@lists.sip-router.org" target="_blank">sr-dev@lists.sip-router.org</a><br>From: <a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a><br>Date: Wed, 28 Oct 2015 14:04:25 +0100<br>Subject: Re: [sr-dev] CAC module idea<div><div class="h5"><br><br>
Hello,<br>
<br>
perhaps piplelimit can be reused to some extent for call attempts.
IIRC, it has also come code for getting some cpu metrics, don't
recall by heart what exactly. But the point would be -- if you are
going to write a cac module, code inside pipelimit can help.<br>
<br>
For available memory, there are some functions to get the available
pkg/shm (used for statistics as well), so this should be easy to
approach. System memory makes sense if there are external libraries
linked by kamailio that potentially consume it, otherwise kamailio
is mailing using pkg/shm for its internal needs. pkg/shm pool have
fixed size.<br>
<br>
Bandwith will be the hardest to assert at SIP layer in my opinion,
but you can take it from a network interface tool. In sip, you can
eventually rely on per codec bandwidth estimation, but that might
not be really accurate and new streams can be added or older removed
via re-INVITE.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
<div>On 28/10/15 13:39, Guillaume wrote:<br>
</div>
<blockquote>
<div dir="ltr">Hi,<br>
<br>
I was thinking about doing a Call Admission Control module. It
will allow the admin to setup limits in order to improve the
global quality and to prevent overload of the service.<br>
It would have these criteria for admitting or not a call (you
can propose other criteria):<br>
* Number of call attempt per second<br>
* Maximal number of concurrent sessions<br>
* CPU limit<br>
* RAM limit<br>
* Maximal bandwidth<br>
* Call priority header<br>
<br>
For now I haven't seen a module for that. And the dialog module
could be useful for to implement the criteria that needs to know
the number of call still active. It doesn't seem to be a too
heavy task and it seems useful. <br>
<br>
What do you think about it?<br>
<br>
<br>
Guillaume<br>
</div>
<br>
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<pre>--
Daniel-Constantin Mierla
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<br></blockquote></div><br><br clear="all"><br>-- <br><div class="gmail_signature">Best regards,<br>Alekzander Spiridonov</div>
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