<div dir="ltr"><div><div><div>Hi,<br><br></div></div>I have reported a bug on rptengine github, for the crash issue: <a href="https://github.com/sipwise/rtpengine/issues/27">https://github.com/sipwise/rtpengine/issues/27</a><br><br></div><div>You mentioned that you have been using rtpengine kamailio module and the rtpengine debian package with success. Was it on ubuntu box or some other linux system? Sorry for asking this, but i am not able move ahead because of this. Any other module suggestions from your end?<br><br></div><div>Thanks  <br></div><div><br></div> <br></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Sep 17, 2014 at 6:45 PM, Abhishek Saini <span dir="ltr"><<a href="mailto:abhishek.saini@enukesoftware.com" target="_blank">abhishek.saini@enukesoftware.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div>Hi Daniel,<br><br></div>Here is something i traced in the log:<br><br>ip-172-31-47-138 rtpengine[4879]: Unknown flag encountered: 'force'<br>ip-172-31-47-138 kernel: [4155571.651074] traps: rtpengine[4884] general protection ip:41e313 sp:7f2bf1934418 error:0 in rtpengine[400000+30000]<br><br></div>What's the cause of this error? i am using code from the master branch. Perhaps this has something to do with the rptengine service crash/termination.<br><br></div>Regards<br></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Sep 17, 2014 at 1:28 PM, Abhishek Saini <span dir="ltr"><<a href="mailto:abhishek.saini@enukesoftware.com" target="_blank">abhishek.saini@enukesoftware.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div><div><div><div><div>Hi Daniel,<br><br></div>As you instructed, i installed kamailio from the master branch (which has rtpengine module). Along with this, i installed the rtpengine package from sipwise, as instructed by them.<br><br></div>I also updated this param : modparam("nathelper", "sipping_from", "<a href="mailto:sip%3Apinger@abc.com" target="_blank">sip:pinger@abc.com</a>") to my domain<br><br></div>Now the scenario is as follows:<br><br></div>1) I am able to call webrtc(firefox and chrome) from iphone, the signalling seems to be working fine, call can be paused, resumed etc.., but there is no audio/video transmission.<br><br></div>2) Still when i call from webrtc to iphone - the retpengine service of ubuntu terminates/crashes (like before) and needs to be restarted.<br><br></div>Does it have any thing to do with rtp port ranges? or is there some other misconfiguration?<br><br><br>Regards,<br></div>Abhishek<br><div><div><div><div><div><div><div><div><div><div><br> <br><div><br></div></div></div></div></div></div></div></div></div></div></div></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Sep 16, 2014 at 6:31 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,<br>
    <br>
    maybe you should play with kamailio master branch (which is in
    testing phase before becoming 4.2)  -- there you have the rtpengine
    -- and see if you get it working. Once that, you can look at using
    an older version, knowing you have it working and be able to
    compare. As I needed latest features, whenever I needed webrtc
    gatewaying, I used devel branch of rtpengine module.<br>
    <br>
    Cheers,<br>
    Daniel<div><div><br>
    <br>
    <div>On 16/09/14 14:24, Abhishek Saini
      wrote:<br>
    </div>
    <blockquote type="cite">
      <div dir="ltr">
        <div>
          <div>
            <div>Hi Daniel,<br>
              <br>
              <br>
            </div>
            I was able to solve a fraction of my problem, Actually, the
            github link had used rtpengine.so and i was using
            rptproxy-ng.so, there is a difference in the flag
            conventions between the two; i modified that to achieve a
            little progress.<br>
            <br>
          </div>
          Now, i am able to call on webrtc(firefox) from sip phone.
          However, after accepting call, there is no audio, and
          disconnecting the call from either end does not disconnect the
          call. <br>
          <br>
        </div>
        When i try to call from webrtc(firefox) to sip phone, there is
        no signalling at all, and the sip phone to webrtc calls can't
        connect after that. (I analyzed that mediaproxy-ng/rtpengine
        process terminates and has to be started again)<br>
        <div>
          <div>
            <div>
              <div><br>
              </div>
              <div>Following are the links to my latest kamailio.cfg
                file and port trace log of sip messages.<br>
                <a href="http://jmp.sh/o0apKgP" target="_blank">http://jmp.sh/o0apKgP</a><br>
                <a href="http://jmp.sh/HXnFRQj" target="_blank">http://jmp.sh/HXnFRQj</a><br>
                <br>
              </div>
              <div>I am clueless at the moment!<br>
              </div>
              <div><br>
              </div>
              <div>Regards,<br>
              </div>
              <div>Abhishek<br>
              </div>
              <div><br>
              </div>
              <div><br>
              </div>
            </div>
          </div>
        </div>
      </div>
      <div class="gmail_extra"><br>
        <div class="gmail_quote">On Tue, Sep 16, 2014 at 1:15 PM,
          Abhishek Saini <span dir="ltr"><<a href="mailto:abhishek.saini@enukesoftware.com" target="_blank">abhishek.saini@enukesoftware.com</a>></span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
            <div dir="ltr">
              <div>
                <div>
                  <div>
                    <div>
                      <div>Hi Daniel,<br>
                        <br>
                      </div>
                      Thanks for this.<br>
                      <br>
                    </div>
                    I took the entire config files and configured it as
                    per my ips and ports, after doing that, still no
                    call establishment(webrtc to classic sip phones and
                    vice-versa). Following is what i get in
                    kamailio.log:<br>
                    <br>
                    rtpp_test(): rtp proxy <udp:<a href="http://127.0.0.1:7722" target="_blank">127.0.0.1:7722</a>>
                    found, support for it enabled<br>
                    ERROR: rtpproxy-ng [rtpproxy.c:1254]:
                    rtpp_function_call(): unknown option ` '<br>
                    ERROR: <script>: ==>
                    duri=[<a>sip:nudg.com:5060;lr;sipml5-outbound;transport=tcp</a>]<br>
                    INFO: <script>: Request coming from WS<br>
                    ERROR: rtpproxy-ng [rtpproxy.c:1254]:
                    rtpp_function_call(): unknown option ` '<br>
                    INFO: <script>: Reply from softphone: 100<br>
                    <br>
                  </div>
                  And this SIP message:<br>
                  SIP/2.0 603 Failed to get local SDP.<br>
                </div>
                <div><br>
                </div>
                Regards,<br>
              </div>
              Abhishek<br>
              <div>
                <div>
                  <div><br>
                    <br>
                    <br>
                  </div>
                </div>
              </div>
            </div>
            <div>
              <div>
                <div class="gmail_extra"><br>
                  <div class="gmail_quote">On Mon, Sep 15, 2014 at 6:19
                    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,<br>
                        <br>
                        the reply code indicates that the media type is
                        not supported, thus there has been no gatewaying
                        between webrtc and classic rtp. Just replacing
                        rtpproxy with rtpengine is not enough, there are
                        different parameters that have to be provided.<br>
                        <br>
                        Searching on web, I see that Carlos has
                        published a config for it, see:<br>
                        - <a href="https://github.com/caruizdiaz/kamailio-ws" target="_blank">https://github.com/caruizdiaz/kamailio-ws</a><br>
                        <br>
                        Cheers,<br>
                        Daniel
                        <div>
                          <div><br>
                            <br>
                            <div>On 15/09/14 12:58, Abhishek Saini
                              wrote:<br>
                            </div>
                            <blockquote type="cite">
                              <div dir="ltr">
                                <div>
                                  <div>Hi, <br>
                                    <br>
                                  </div>
                                  I have successfully setup rtpproxy-ng
                                  kamailio module and mediaproxy-ng
                                  package on my ubuntu box. As suggested
                                  here:<br>
                                  <a href="http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html" target="_blank">http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html</a><br>
                                </div>
                                <div><br>
                                </div>
                                <div>I have kept rtpproxy-ng's
                                  configuration same as the rtpproxy
                                  module, but still not able to connect
                                  the webrtc calls to classic sip phones
                                  (and vice-versa). Below is the sip
                                  message that is traced:<br>
                                  <br>
                                  <br>
                                  SIP/2.0 488 Not acceptable here.<br>
                                  Via: SIP/2.0/TCP
                                  54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$<br>
                                  Via: SIP/2.0/WS
                                  df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$<br>
                                  From: "admin" <<a href="mailto:sip%3Aadmin@abc.com" target="_blank">sip:admin@abc.com</a>>;tag=bzhwwG8nT2gFwwJgIyrz.<br>
                                  To: <<a href="mailto:sip%3Ahari@abc.com" target="_blank">sip:hari@abc.com</a>>;tag=OIllTQf.<br>
                                  Call-ID:
                                  31464f04-27e6-b11c-3a63-ba1d4d2d4d5a.<br>
                                  CSeq: 65463 INVITE.<br>
                                  User-Agent: LinphoneIPhone/2.2.1
                                  (belle-sip/1.3.2).<br>
                                  Supported: replaces, outbound.<br>
                                  Content-Length: 0.<br>
                                  <br>
                                </div>
                                <div>Can you please let me know, what's
                                  going wrong and how can i proceed.<br>
                                  <br>
                                </div>
                                <div>Regards,<br>
                                </div>
                                <div>Abhishek<br>
                                </div>
                                <div><br>
                                  <br>
                                </div>
                                <div><br>
                                   <br>
                                </div>
                              </div>
                            </blockquote>
                            <br>
                          </div>
                        </div>
                        <span>
                          <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>
                        </span></div>
                    </blockquote>
                  </div>
                  <br>
                </div>
              </div>
            </div>
          </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>
  </div></div></div>

</blockquote></div><br></div>
</div></div></blockquote></div><br></div>
</div></div></blockquote></div><br></div>