<div dir="ltr">Here is it<br><a href="http://pastebin.com/JkkM4M5m">http://pastebin.com/JkkM4M5m</a><br></div><div class="gmail_extra"><br><div class="gmail_quote">2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla <span dir="ltr"><<a 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">
  
    
  
  <div bgcolor="#FFFFFF" text="#000000">
    There are no major changes in 4.3 comparing with 4.2 in regards to
    websocket -- the implementation is quite mature for a long time.<br>
    <br>
    Looks like websocket connection is not available. Can you look at
    javascript debug console in the browser to see what is printing?<span class="HOEnZb"><font color="#888888"><br>
    <br>
    Daniel</font></span><div><div class="h5"><br>
    <br>
    <div>On 23/06/15 17:23, Alexandru Covalschi
      wrote:<br>
    </div>
    <blockquote type="cite">
      <div dir="ltr">
        <div>without fix_nated_contact error behaviour is the same<br>
        </div>
        maybe I should upgrade to 4.3 ?<br>
      </div>
      <div class="gmail_extra"><br>
        <div class="gmail_quote">2015-06-23 14:08 GMT+03:00 Alexandru
          Covalschi <span dir="ltr"><<a href="mailto:568691@gmail.com" target="_blank">568691@gmail.com</a>></span>:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
            <div dir="ltr">Here's the trace on port which I use for ws
              server. Don't look at fix_nated_contact, I'll fix later -
              now the trouble is that Kamailio can't establish a ws
              connection properly. Client is SIPML5 demo phone<br>
              <a href="http://pastebin.com/LvAk2HkP" target="_blank">http://pastebin.com/LvAk2HkP</a><br>
            </div>
            <div>
              <div>
                <div class="gmail_extra"><br>
                  <div class="gmail_quote">2015-06-23 14:03 GMT+03:00
                    Alexandru Covalschi <span dir="ltr"><<a href="mailto:568691@gmail.com" target="_blank">568691@gmail.com</a>></span>:<br>
                    <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                      <div dir="ltr">I solved the SIP voice trouble, but
                        WebRTC problem still exists. What kind of trace
                        I must do to make my post more informative?<br>
                      </div>
                      <div>
                        <div>
                          <div class="gmail_extra"><br>
                            <div class="gmail_quote">2015-06-23 10:46
                              GMT+03:00 Daniel-Constantin Mierla <span dir="ltr"><<a 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">
                                <div bgcolor="#FFFFFF" text="#000000">
                                  Hello,<span><br>
                                    <br>
                                    <div>On 23/06/15 04:10, Alexandru
                                      Covalschi wrote:<br>
                                    </div>
                                    <blockquote type="cite">
                                      <div dir="ltr">
                                        <div>Hello. I'm trying to set up
                                          this (v 4.2 stable):<br>
                                        </div>
                                        peer <--> ec2
                                        <--kamailio+rtpengine-->
                                        asterisk<br clear="all">
                                        <div>
                                          <div>
                                            <div>scheme<br>
                                              <br>
                                            </div>
                                            <div>I use advertised adress
                                              for SIP and WS
                                              connections.<br>
                                            </div>
                                            <div>The problem is that on
                                              SIP I get one way audio -
                                              I can receive audio from
                                              asterisk, but I can't
                                              transmit audio there - my
                                              SIP UA tries to send data
                                              to Kamailio-s local EC2
                                              IP.</div>
                                          </div>
                                        </div>
                                      </div>
                                    </blockquote>
                                    <br>
                                  </span> you should grab a ngrep trace
                                  on server to see what happens in the
                                  signaling in order to be able to
                                  provide some hints on solving it.<br>
                                  <br>
                                  Cheers,<br>
                                  Daniel<br>
                                  <br>
                                  <blockquote type="cite"><span>
                                      <div dir="ltr">
                                        <div>
                                          <div>
                                            <div> In case of WebRTC I
                                              get lot's of erros:<br>
                                              <br>
                                              Jun 23 01:58:57 kamailio
                                              /usr/sbin/kamailio[18325]:
                                              WARNING: <core>
                                              [msg_translator.c:2778]:
                                              via_builder(): TCP/TLS
                                              connection (id: 0) for
                                              WebSocket could not be
                                              found<br>
                                              Jun 23 01:58:57 kamailio
                                              /usr/sbin/kamailio[18325]:
                                              ERROR: <core>
                                              [msg_translator.c:1996]:
                                              build_req_buf_from_sip_req():
                                              could not create Via
                                              header<br>
                                              Jun 23 01:58:57 kamailio
                                              /usr/sbin/kamailio[18325]:
                                              ERROR: <core>
                                              [forward.c:584]:
                                              forward_request():
                                              building failed<br>
                                              Jun 23 01:58:57 kamailio
                                              /usr/sbin/kamailio[18325]:
                                              ERROR: sl
                                              [sl_funcs.c:387]:
                                              sl_reply_error(): ERROR:
                                              sl_reply_error used: I'm
                                              terribly sorry, server
                                              error occurred (1/SL)<br>
                                              <br>
                                            </div>
                                            <div>The call reaches
                                              Asterisk, but not
                                              vice-versa. No media is
                                              being transferred.<br>
                                              <br>
                                            </div>
                                            <div>Rtpengine flags I use:<br>
                                            </div>
                                            <div>For SIP: 
                                              rtpengine_manage("trust-adress
                                              replace-origin
                                              replace-session-connection
                                              RTP/AVP");<br>
                                            </div>
                                            <div>For WS: 
                                              rtpengine_manage("trust-address
                                              replace-origin
                                              replace-session-connection
                                              ICE=force RTP/AVP");<br>
                                              <br>
                                            </div>
                                            <div>Do you have any ideas
                                              how ti fix that? I also
                                              make REGFWD's to Asterisk<br>
                                            </div>
                                            <div>-- <br>
                                              <div>
                                                <div dir="ltr">Alexandru
                                                  Covalschi<br>
                                                  ABRISS-Solutions
                                                  <div>VoIP engineer and
                                                    system administrator<br>
                                                    phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>
                                                    web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div>
                                                </div>
                                              </div>
                                            </div>
                                          </div>
                                        </div>
                                      </div>
                                      <br>
                                      <fieldset></fieldset>
                                      <br>
                                    </span>
                                    <pre>_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a>
<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><span><font color="#888888">
</font></span></pre>
                                    <span><font color="#888888"> </font></span></blockquote>
                                  <span><font color="#888888"> <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>
Book: SIP Routing With Kamailio - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a></pre>
                                    </font></span></div>
                                <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" rel="noreferrer" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
                                <br>
                              </blockquote>
                            </div>
                            <br>
                            <br clear="all">
                            <br>
                            -- <br>
                            <div>
                              <div dir="ltr">Alexandru Covalschi<br>
                                ABRISS-Solutions
                                <div>VoIP engineer and system
                                  administrator<br>
                                  phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>
                                  web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div>
                              </div>
                            </div>
                          </div>
                        </div>
                      </div>
                    </blockquote>
                  </div>
                  <br>
                  <br clear="all">
                  <br>
                  -- <br>
                  <div>
                    <div dir="ltr">Alexandru Covalschi<br>
                      ABRISS-Solutions
                      <div>VoIP engineer and system administrator<br>
                        phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>
                        web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div>
                    </div>
                  </div>
                </div>
              </div>
            </div>
          </blockquote>
        </div>
        <br>
        <br clear="all">
        <br>
        -- <br>
        <div>
          <div dir="ltr">Alexandru Covalschi<br>
            ABRISS-Solutions
            <div>VoIP engineer and system administrator<br>
              phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>
              web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div>
          </div>
        </div>
      </div>
      <br>
      <fieldset></fieldset>
      <br>
      <pre>_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a>
<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>
</pre>
    </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>
Book: SIP Routing With Kamailio - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a></pre>
  </div></div></div>

<br>_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
<a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" rel="noreferrer" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
<br></blockquote></div><br><br clear="all"><br>-- <br><div class="gmail_signature"><div dir="ltr">Alexandru Covalschi<br>ABRISS-Solutions<div>VoIP engineer and system administrator<br>phone: +37367398493<br>web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div></div></div>
</div>