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    <p>Hello,</p>
    <p>yes call record_route function here:</p>
    <p># record routing<br>
              if (!is_method("REGISTER|MESSAGE"))<br>
                      record_route();</p>
    <p>and fix_nated_contact here:</p>
    <p>onreply_route[handle_nat] {<br>
              if (nat_uac_test("private-contact")) {<br>
                      fix_nated_contact();<br>
              }<br>
              if ($socket_in =~ "wss") {<br>
                      fix_nated_contact();<br>
              }<br>
              if (has_body_part("application/sdp") &&
      t_check_status("200")) {<br>
              route(RTPENGINE);<br>
              }<br>
      }<br>
    </p>
    <p>Regards<br>
    </p>
    <div class="moz-signature"><br>
      <img src="cid:part1.o2v1wZX9.QKZXnHjC@mesaproyectos.com"
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    <div class="moz-cite-prefix">El 6/02/2025 a las 8:11 a. m., Răzvan
      Crainea escribió:<br>
    </div>
    <blockquote type="cite"
      cite="mid:fc10ebde-1034-4db8-9ace-7e38f3da6727@opensips.org">Hello!
      <br>
      <br>
      Are you calling record_route? Also, make sure you call
      fix_nated_contact() on the 200 OK. Read this blog post for more
      information:
      <br>
<a class="moz-txt-link-freetext" href="https://blog.opensips.org/2017/02/22/troubleshooting-missing-ack-in-sip/">https://blog.opensips.org/2017/02/22/troubleshooting-missing-ack-in-sip/</a>
      <br>
      <br>
      Best regards,
      <br>
      <br>
      Răzvan Crainea
      <br>
      OpenSIPS Core Developer / SIPhub CTO
      <br>
      <a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a> / <a class="moz-txt-link-freetext" href="https://www.siphub.com">https://www.siphub.com</a>
      <br>
      <br>
      On 1/31/25 3:55 PM, VoIP via Users wrote:
      <br>
      <blockquote type="cite">Good morning everyone,
        <br>
        <br>
        I'm trying to implement this type of scenario:
        <br>
        <br>
        WSS -> load_balancer -> UDP Gateway (Asterisk)
        <br>
        <br>
        Everything works up to the 200 OK received from the gateway and
        forwarded from OpenSIPs to the WebRTC clients.
        <br>
        <br>
        I don't see the ACK sent from the WebRTC client to OpenSIPs to
        commit the 200OK.
        <br>
        <br>
        WebRTC -> UDP and UDP -> WebRTC calls between users work
        correctly and analyzing the 200 OK of a call between users and a
        call via load_balancer, the truth is that I do not find
        differences that justify this type of error.
        <br>
        <br>
        I'm writing a tutorial, for now in Spanish, dedicated to the
        subject but without that piece I can't finish it.
        <br>
        <br>
        Thank you in advance for the help
        <br>
        <br>
      </blockquote>
      <br>
      <br>
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      <br>
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      <br>
    </blockquote>
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