<div dir="ltr"><div>Hi,<br><br>Please find attached the output of ngrep for three type of combinations/connections:<br><br></div>key: Blink is the desktop sip client and ntw means network.<br><div><br>blink2blink_same_ntw_successful<br>

webrtc2blink_same_ntw_failed<br>webrtc2webrtc_same_ntw_successful<br></div><div><br>We also need to enable webrtc to classic sip phone calls, like on iphones/desktops etc. I could not find a good tutorial on rtpengine, and the steps to replace rtpproxy with rtpengine.<br>

<br></div><div>Please suggest me on this. <br><br></div><div>Regards,<br>Abhishek<br></div><div><br> <br></div><div><br></div><div><br><br><br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Sep 4, 2014 at 6:02 PM, Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>></span> wrote:<br>

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    Hello,<br>
    <br>
    maybe you can send to mailing list the output of ngrep so we can
    look and check if a rtp relay is used.<br>
    <br>
    If you need to bridge webrtc to classic sip phone, you have to use
    rtpengine.<br>
    <br>
    Cheers,<br>
    Daniel<div><div><br>
    <br>
    <div>On 04/09/14 13:01, Abhishek Saini
      wrote:<br>
    </div>
    <blockquote type="cite">
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                    <div>Hi Daniel,<br>
                      <br>
                    </div>
                    Thanks, i was able to use the command you provided,
                    but did not find the chunks you have
                    specified(a=nortproxy:yes (iirc)) in the data.
                    Checked by calling from webrtc client to a desktop
                    client(blink).<br>
                    <br>
                  </div>
                  When is rtpproxy used though? Kamailio says that it
                  only transmits SIP signals and has not much to do with
                  the media(voice or video). So, that means, it utilizes
                  the rtpproxy to transmit the SIP signals(for
                  non-symmetric NAT), If so then i think, the rtpproxy
                  is working fine, as i have always been able to make
                  and receive calls and only the media (voice or video)
                  are not working (cross network).<br>
                  <br>
                </div>
                I have also setup webrtc - it's working fine (firefox to
                firefox) but when i call from firefox to desktop client,
                it does not work(only rings, but does not connect). <br>
              </div>
              I read about webrtc_breaker but there does not seem to be
              a module for that in kamailio.<br>
              <br>
            </div>
            I think these two issues are somehow interlinked, please
            suggest me on this. <br>
            <br>
          </div>
          Regards,<br>
        </div>
        Abhishek <br>
      </div>
      <div class="gmail_extra"><br>
        <br>
        <div class="gmail_quote">On Thu, Sep 4, 2014 at 1:28 PM,
          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">
            <div bgcolor="#FFFFFF" text="#000000"> Hello,
              <div><br>
                <br>
                <div>On 04/09/14 09:20, Abhishek Saini wrote:<br>
                </div>
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                            <div>Hi Daniel,<br>
                              <br>
                            </div>
                            Thanks for reply.<br>
                            <br>
                          </div>
                          I did install patched rtpproxy and did
                          configure it the way you have described
                          (advertising address - found that after
                          posting the comment). But it still does not
                          seem to work.<br>
                          <br>
                        </div>
                        I don't quite know how can i debug, if rtpproxy
                        is actually being used.<br>
                      </div>
                    </div>
                  </div>
                </blockquote>
              </div>
              use ngrep to look at sip traffic, like:<br>
              <br>
              ngrep -d any -qt -W byline port 5060<br>
              <br>
              If rtpproxy was enforced, you should see a=nortproxy:yes
              (iirc) in the SDP. Also, the media IP in SDP should change
              from incoming INVITE to what is sent out in the IP of
              rtpproxy.<br>
              <br>
              Cheers,<br>
              Daniel
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                        <div><br>
                        </div>
                        Regards,<br>
                      </div>
                      Abhishek<br>
                      <div><br>
                      </div>
                    </div>
                    <div class="gmail_extra"><br>
                      <br>
                      <div class="gmail_quote">On Thu, Sep 4, 2014 at
                        12:34 PM, 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">Hello,<br>
                          <br>
                          no time to look at config, but if you run the
                          sip server on a private IP behind a port
                          forwarding address, you have to use also
                          rtpproxy with advertising address -- see the
                          second parameter of rtpproxy_manage() or
                          search on the web for a patch to rtpproxy to
                          add advertising address via command line
                          parameter.<br>
                          <br>
                          Cheers,<br>
                          Daniel
                          <div>
                            <div><br>
                              <br>
                              On 03/09/14 12:23, Abhishek Saini wrote:<br>
                              <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"> Hi,<br>
                                <br>
                                I have setup kamailio 4.1.0 on an EC2
                                xlarge instance. The voice and video
                                calls seem to work well when both the
                                devices are connected to the same
                                network, however, when one device
                                connects to a different network (the two
                                devices now are on different networks),
                                they are able to register on SIP server,
                                and even call can be triggered and
                                accepted between the two devices but
                                there is no video/audio transmission.<br>
                                <br>
                                I have setup rtpproxy but i don't know
                                whether it's working or not.<br>
                                <br>
                                Any help on this would be highly
                                appreciated.<br>
                                <br>
                                <br>
                                Following is my kamailio.cfg file:<br>
                              </blockquote>
                              <br>
                            </div>
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                          <span><font color="#888888"> -- <br>
                              Daniel-Constantin Mierla<br>
                              <a href="http://twitter.com/#%21/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><br>
                              Next Kamailio Advanced Trainings 2014 - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a><br>
                              Sep 22-25, Berlin, Germany<br>
                              <br>
                              <br>
_______________________________________________<br>
                              SIP Express Router (SER) and Kamailio
                              (OpenSER) - sr-users mailing list<br>
                              <a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a><br>
                              <a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
                            </font></span></blockquote>
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                      <br>
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                  <br>
                  <pre cols="72">-- 
Daniel-Constantin Mierla
<a href="http://twitter.com/#%21/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>
Next Kamailio Advanced Trainings 2014 - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a>
Sep 22-25, Berlin, Germany</pre>
                </div>
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          </blockquote>
        </div>
        <br>
      </div>
    </blockquote>
    <br>
    <pre cols="72">-- 
Daniel-Constantin Mierla
<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>
Next Kamailio Advanced Trainings 2014 - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a>
Sep 22-25, Berlin, Germany</pre>
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