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    <div class="moz-cite-prefix">On 17/09/14 17:19, PaweÅ‚ Sternal wrote:<br>
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cite="mid:CAHexhxLN1hw2R_MriAjNbPrgJWARX3RyKQa_0wbWzLTO7BsjuA@mail.gmail.com"
      type="cite">
      <div dir="ltr">Thanks Daniel, you save my day :-) </div>
    </blockquote>
    You're welcome! Great to see you ended up with more than expected
    :-)<br>
    <br>
    Cheers,<br>
    Daniel<br>
    <br>
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cite="mid:CAHexhxLN1hw2R_MriAjNbPrgJWARX3RyKQa_0wbWzLTO7BsjuA@mail.gmail.com"
      type="cite">
      <div dir="ltr">Two rtpengine solves my problem and works perfect.
        This solution also adds me the possibility to record calls,
        when between two rtpengine instances I will put rtpproxy.
        <div><br>
        </div>
        <div>
          <div>Regards,</div>
          <div>Pawel</div>
        </div>
      </div>
      <div class="gmail_extra"><br>
        <div class="gmail_quote">2014-09-17 9:35 GMT+02:00
          Daniel-Constantin Mierla <span dir="ltr"><<a
              moz-do-not-send="true" href="mailto:miconda@gmail.com"
              target="_blank">miconda@gmail.com</a>></span>:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">Hello,<br>
            <br>
            I would add the RTP-WebRTC gateway between SIP Kamailio and
            SIP UAC, from resources point of view, it is the only leg
            that needs encryption/decryption.<br>
            <br>
            Otherwise, you can try to work with two rtpengine instances
            (sets) in WS Kamailio, one to use for ws client to proxy and
            the other one for the leg from proxy to ws client. It will
            be a communication between them with classic rtp, both
            having towards ws client webrtc. It has the drawback of
            decryption and encryption done two times for the same call.
            You would need to add rtpengine set id in record-route to be
            able to handle properly the re-INVITE, BYE, etc.<br>
            <br>
            Another option that I would use is to send a negative reply
            from SIP kamailio, catch that in failure_route in WS
            Kamailio and engace there the rtpengine with proper flags.
            E.g., you assume it is going to be webrtc-to-webrtc, so no
            encryption/decryption added first time invite comes from WS
            client. You forward to SIP kamailio, which based on
            location, if it discovers that the callee is classic
            SIP-RTP, will send a 4xx back to WS Kamailio -- you end
            previous rtpengine session and engage it again with new
            flags (use branch route for managing rtpengine -- like it is
            done in default kamailio.cfg for rtpproxy).<br>
            <br>
            Cheers,<br>
            Daniel
            <div>
              <div class="h5"><br>
                <br>
                On 15/09/14 20:30, PaweÅ‚ Sternal wrote:<br>
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              .8ex;border-left:1px #ccc solid;padding-left:1ex">
              <div>
                <div class="h5">
                  Hi. Another topic about WebRTC, websockets with
                  kamailio and rtpengine ;-)<br>
                  <br>
                  My problem is how to distinguish a call to WS UAC and
                  how to SIP UAC in scenarios:<br>
                  <br>
                  1) WS client1 -> WS kamailio -> SIP kamailio
                  -> SIP UAC<br>
                  <br>
                  2) WS client1 -> WS kamailio -> SIP kamailio
                  -> WS kamailio -> WS client2<br>
                  <br>
                  WS kamailio is a proxy, SIP kamailio is a registrar<br>
                  <br>
                  When "WS client1" is calling to "123123" WS kamailio
                  doesn't know if "123123" was registered from "WS
                  client2" or from SIP UAC.<br>
                  <br>
                  I have in this case rtpengine_manage(".......
                  RTP/AVP"), but when INVITE is returned to WS kamailio?
                  RTP/SAVPF?<br>
                  <br>
                  Probably it is obvious, however...<br>
                  <br>
                  When WS client2 reply with 200OK,
                  rtpengine_manage("..... ICE=force") to WS client1 SDP
                  is sent without a:fingerprint. sipml5 dumps warning:<br>
                  <br>
                  message: "Could not negotiate answer SDP; cause =
                  NO_DTLS_FINGERPRINT<br>
                  <br>
                  I tried different combinations... and I'm stuck :/<br>
                  <br>
                  Regards<br>
                  <br>
                </div>
              </div>
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                class="HOEnZb"><font color="#888888"><br>
                </font></span></blockquote>
            <span class="HOEnZb"><font color="#888888">
                <br>
                -- <br>
                Daniel-Constantin Mierla<br>
                <a moz-do-not-send="true"
                  href="http://twitter.com/#%21/miconda" target="_blank">http://twitter.com/#!/miconda</a>
                - <a moz-do-not-send="true"
                  href="http://www.linkedin.com/in/miconda"
                  target="_blank">http://www.linkedin.com/in/miconda</a><br>
                Next Kamailio Advanced Trainings 2014 - <a
                  moz-do-not-send="true" href="http://www.asipto.com"
                  target="_blank">http://www.asipto.com</a><br>
                Sep 22-25, Berlin, Germany<br>
                <br>
              </font></span></blockquote>
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    <br>
    <pre class="moz-signature" cols="72">-- 
Daniel-Constantin Mierla
<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>
Next Kamailio Advanced Trainings 2014 - <a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a>
Sep 22-25, Berlin, Germany</pre>
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