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    <div class="moz-cite-prefix">Hi,<br>
      <br>
      The problem is that FreeSWITCH doesn't forward Late SDP
      negotiation in a RE-INVITE scenario which end up to one way audio.<br>
      <br>
      &nbsp;After testing the option "enable-soa" didn't resolved the issue.<br>
      <br>
      I want to bridge two calls that are already established (I am not
      dealing with INVITE but with Re-INVITE) .<br>
      That's why inbound-late-negotiation (which is within the dialplan)
      is not in the scope of this (I have it activated already).<br>
      <br>
      <div style="">&gt; When Call B is answered, you should use SDP in
        200 OK as basis for SDP of ACK in Call A. Otherwise you will get
        one way audio as expect.<br>
        <br>
        Yes. This is the scenario I described at state 4) of the schema.<br>
      </div>
      <br>
      Thanks<br>
      <br>
      BNML<br>
      <br>
      Le 06/28/2013 07:00 PM, Muhammad Shahzad a &eacute;crit&nbsp;:<br>
    </div>
    <blockquote
cite="mid:CAFZQphwr7eicBCo5Fn67QbPphJG=9adMNQ1DZ3G3f2YyrFr+YQ@mail.gmail.com"
      type="cite">
      <div dir="ltr">I am not sure what you are trying to do here. You
        said we want to originate two calls from your PABX to carrier
        through FS, i.e.
        <div><br>
        </div>
        <div style="">Call A: PABX -&gt; FS -&gt; Carrier -&gt; User A</div>
        <div style="">Call B: PABX -&gt; FS -&gt; Carrier -&gt; User B</div>
        <div style=""><br>
        </div>
        <div style="">And then bridge them on FS, such that,</div>
        <div style=""><br>
        </div>
        <div style="">User A &lt;- Carrier &lt;- FS -&gt; Carrier -&gt;
          User B</div>
        <div style=""><br>
        </div>
        <div style="">I think there are easier ways to achieve this. But
          anyway in this scenario,</div>
        <div style=""><br>
        </div>
        <div style="">Since Call A does not have SDP, which means you
          are triggering Late SDP Negotiation. Make sure you have
          enabled in sofia profile,</div>
        <div style=""><br>
        </div>
        <div style="">
          <pre style="padding:1em;border:1px dashed rgb(47,111,171);color:rgb(0,0,0);background-color:rgb(249,249,249);line-height:1.1em">&lt;param name="inbound-late-negotiation" value="true"/&gt;</pre>
        </div>
        <div style=""><br>
        </div>
        <div style="">So, that FS does not Answer it with local SDP.
          Additionally you may need to disable SOA,</div>
        <div style=""><br>
        </div>
        <div style="">
          <pre style="padding:1em;border:1px dashed rgb(47,111,171);color:rgb(0,0,0);background-color:rgb(249,249,249);line-height:1.1em">&lt;param name="enable-soa" value="false"/&gt;</pre>
        </div>
        <div style=""><br>
        </div>
        <div style="">When 200 OK for Call A arrives (from carrier,
          through FS to PABX), you use it as basis for SDP of INVITE in
          Call B. That's interesting..</div>
        <div style=""><br>
        </div>
        <div style="">When Call B is answered, you should use SDP in 200
          OK as basis for SDP of ACK in Call A. Otherwise you will get
          one way audio as expect.</div>
        <div style=""><br>
        </div>
        <div style=""><br>
        </div>
        <div style=""> Thank you.</div>
        <div style=""><br>
        </div>
        <div style=""><br>
        </div>
      </div>
      <div class="gmail_extra"><br>
        <br>
        <div class="gmail_quote">On Fri, Jun 28, 2013 at 6:56 PM, BNML <span
            dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:bnml@andrexen.com" target="_blank">bnml@andrexen.com</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi all,<br>
            <br>
            I am having trouble with a transfert failure in a special
            scenario which<br>
            i wasn't able to find documentation or configuration option
            solving it.<br>
            <br>
            I have a PBX creating two outbound call, it want to bridge
            them on the<br>
            FreeSWITCH.<br>
            It is sending a first INVITE without SDP and it will send
            the SDP in the<br>
            ACK.<br>
            <br>
            (This is not breaking RFC3261 but I admit rarely seen
            scenario.<br>
            <a moz-do-not-send="true"
              href="http://tools.ietf.org/html/rfc3261#section-13.2.1"
              target="_blank">http://tools.ietf.org/html/rfc3261#section-13.2.1</a>
            : "The UAC MAY choose<br>
            to add a message body to the INVITE.". Looking at this
            scenario there<br>
            is, indeed, at least one interesting scenario where it can
            helps<br>
            avoiding creating another useless Re-INVITE transaction, see
            below)<br>
            <br>
            For the second call it is sending a re-invite with SDP.<br>
            FreeSWITCH will understand and forward (because of
            "bypass_media") the<br>
            second Re-INVITE but does never forward the first Re-INVITE
            (The one<br>
            without SDP in the INVITE but in the ACK) to carrier, so
            this will end<br>
            up in one-way-audio.<br>
            <br>
            A little schema to help :<br>
            <br>
            PBX &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;FREESWITCH &nbsp; &nbsp; &nbsp; COMMENT<br>
            1) INVITE (a-leg) &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; [THERE IS NO SDP]<br>
            2) &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 200 OK (fs-a-leg)[FreeSWITCH answer
            with the SDP<br>
            information before any media update occurs, but my PBX is
            now aware of<br>
            media informations]<br>
            3) INVITE+SDP (b-leg) &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; [This is b-leg
            updated media<br>
            information with previous 200 OK (fs-a-leg)]<br>
            4) &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 200 OK (fs-b-leg)[My PBX updated
            media and has<br>
            b-leg freeswitch media information that it will send in the
            ACK+SDP to<br>
            finalize the both-legs Re-INVITE]<br>
            5) ACK (b-leg) &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; [Acknoledgement of
            b-leg<br>
            Re-INVITE transaction]<br>
            6) ACK+SDP (a-leg) &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; [ *** problem ***
            it assume<br>
            FreeSWITCH will update media information and forward to the
            carrier<br>
            which it won't]<br>
            <br>
            The solution i imagine is FreeSWITCH SHOULD trigger a
            Re-INVITE either<br>
            for the a-leg at INVITE (without SDP in "1") receiption (and
            update<br>
            media information when receiving the ACK+SDP) _OR_ create a
            Re-INVITE<br>
            transaction at "6" when receiving ACK+SDP. One way or
            another, I am<br>
            supposing FreeSWITCH refuse to forward the INVITE without
            SDP at first<br>
            maybe because of carrier compatibility ? If not, this seems
            the way to go.<br>
            <br>
            Has anyone faced such a scenario ? Am I rigth assuming a
            patch is needed<br>
            ? If yes, any suggestions are welcome.<br>
            <br>
            Thanks you<br>
            <br>
            BNML<br>
            <br>
            <br>
_________________________________________________________________________<br>
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          </blockquote>
        </div>
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        <div><br>
        </div>
        -- <br>
        <div><span
style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">Mit

            freundlichen Gr&uuml;&szlig;en</span></div>
        <span
style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">Muhammad

          Shahzad</span><br
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        <span
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        <span
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        <span
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        <span
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          176 99 83 10 85</span><br
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      <br>
      <pre wrap="">_________________________________________________________________________
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</pre>
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