<div dir="ltr">I must find out how it is with licence of school project but of course if it wouldn't be problem I can share git repo. Just today I asked in another thread here on [sr-dev] if there is way to use internal udp socket to send packets to VoIP client in my module.<br>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Apr 16, 2014 at 8:57 AM, Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
  
    
  
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    <br>
    <div>On 14/04/14 10:29, Cock Ootec wrote:<br>
    </div>
    <blockquote type="cite">
      <div dir="ltr">I am sorry forĀ 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></div>
    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 moreĀ  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<div><div class="h5"><br>
    <br>
    <blockquote 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 href="mailto:oej@edvina.net" target="_blank">oej@edvina.net</a>></span>
          wrote:<br>
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            <div><br>
              On 07 Apr 2014, at 15:39, Andreas Granig <<a href="mailto:agranig@sipwise.com" target="_blank">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><font color="#888888"><br>
                /O<br>
              </font></span>
            <div>
              <div>><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>
                > _______________________________________________<br>
                > sr-dev mailing list<br>
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                <br>
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              </div>
            </div>
          </blockquote>
        </div>
        <br>
      </div>
      <br>
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</pre>
    </blockquote>
    <br>
    </div></div><span class="HOEnZb"><font color="#888888"><pre cols="72">-- 
Daniel-Constantin Mierla - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a>
<a href="http://twitter.com/#!/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a></pre>
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