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<div class="moz-cite-prefix">On 14/04/14 10:29, Cock Ootec wrote:<br>
</div>
<blockquote
cite="mid:CAJetf70WKQDp=Ld3+k_cYnnQFeeXkRUpoaGkeCgCcVc95FKkEQ@mail.gmail.com"
type="cite">
<div dir="ltr">I am sorry for inconvenience. Yes I asked these
questions in developer context.
<div>Now I am able to work with RTP packets in my module (I know
that this seems to be useless</div>
</div>
</blockquote>
It is not useless if needed. Maybe you can share more details and we
can give hints on what could be reused for easier development. Also,
you may be surprised to find other people having same interest and
in case it is something wanted by more and you want to release your
code open source, then we can include the module in official
kamailio git repository.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
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cite="mid:CAJetf70WKQDp=Ld3+k_cYnnQFeeXkRUpoaGkeCgCcVc95FKkEQ@mail.gmail.com"
type="cite">
<div dir="ltr">
<div> but it is for my school project ;-) ) so if anyone asks it
is possible.<br>
<br>
Thanks for your help</div>
</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On Mon, Apr 7, 2014 at 3:41 PM, Olle E.
Johansson <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:oej@edvina.net" target="_blank">oej@edvina.net</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class=""><br>
On 07 Apr 2014, at 15:39, Andreas Granig <<a
moz-do-not-send="true" href="mailto:agranig@sipwise.com">agranig@sipwise.com</a>>
wrote:<br>
<br>
> Hi,<br>
><br>
> On 03/26/2014 03:18 PM, Alex Balashov wrote:<br>
>> No. You can't route the RTP and RTCP traffic to
Kamailio, by definition.<br>
>><br>
>> You keep asking questions that betray a lack of
basic understanding of SIP network elements. I think you
should take Olle's suggestion and learn how it works.<br>
><br>
> For the sake of discussion, I think it's somewhat
possible to route<br>
> rtp/rtcp with kamailio. Does it make sense? No. Would
it work? Probably,<br>
> in a limited way.<br>
</div>
Of course, if you are a developer, you can do anything. :-)<br>
<br>
But the question wasn't asked in the developer context, at
least I did not<br>
parse it that way...<br>
<span class="HOEnZb"><font color="#888888"><br>
/O<br>
</font></span>
<div class="HOEnZb">
<div class="h5">><br>
> So if I wanted to do something like this, then I'd
find the point where<br>
> kamailio is actually calling recv(), then find out
where it feeds the<br>
> received data into the sip parser. There, I'd
implement the logic to<br>
> quickly check if what we're dealing with is an rtp
packet, and handle it<br>
> differently than other packets. For SDP in request
and response bodies<br>
> flowing through my config, I'd modify SDP to put
5060 as media port for<br>
> the various streams.<br>
><br>
> Now since every packet will be received on port
5060, you can't really<br>
> distinguish between different streams, as you can't
rely on the source<br>
> address advertised in SDP because of NAT, so any
NAT scenario with more<br>
> than one phone behind that NAT is going to break
the whole thing. Well,<br>
> putting aside NAT, you now would have to maintain
mapping tables of<br>
> source addresses announced in SDP and check (and
rely on) them for<br>
> inbound packets and map them to the outbound leg
based on the source<br>
> address. That might work for non-NAT scenarios (but
who's using NAT in a<br>
> world of IPv6 anyways?).<br>
><br>
> Now the question is, why would anyone want to do
that? If the intention<br>
> is to make it work better in NAT environments, then
our OP has probably<br>
> not thought it through entirely.<br>
><br>
> Andreas<br>
><br>
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<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla - <a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a>
<a class="moz-txt-link-freetext" href="http://twitter.com/#!/miconda">http://twitter.com/#!/miconda</a> - <a class="moz-txt-link-freetext" href="http://www.linkedin.com/in/miconda">http://www.linkedin.com/in/miconda</a></pre>
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