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    <div class="moz-cite-prefix">I am suspecting a patch on mod_sofia
      migth be needed in order to resolve the Late SDP Negotiation in an
      RE-INVITE scenario to avoir the one-way audio. I can get involved
      in writing the patch, if needed.<br>
      <br>
      Here is the FS console logs along with sip trace <br>
      <a class="moz-txt-link-freetext" href="http://pastebin.freeswitch.org/21141">http://pastebin.freeswitch.org/21141</a><br>
      <br>
      Le 01/07/2013 14:40, Muhammad Shahzad a &eacute;crit&nbsp;:<br>
    </div>
    <blockquote
cite="mid:CAFZQphxAQjpYT8jXMkKb0g6pX5Fm=F+4HnCULNNCSajZcFcEXw@mail.gmail.com"
      type="cite">
      <div dir="ltr">Can you send FS console logs along with sip trace
        enabled? Just curious how FS responds to this situation.
        <div><br>
        </div>
        <div>Thank you.</div>
      </div>
      <div class="gmail_extra"><br>
        <br>
        <div class="gmail_quote">On Mon, Jul 1, 2013 at 7:21 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">
            <div bgcolor="#FFFFFF" text="#000000">
              <div>
                <div class="im">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>
                <div>
                  <div class="im">&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>
                  </div>
                  <div class="im"> Yes. This is the scenario I described
                    at state 4) of the schema.<br>
                  </div>
                </div>
                <div class="im"> <br>
                  Thanks<br>
                  <br>
                  BNML<br>
                  <br>
                  Le 06/28/2013 07:00 PM, Muhammad Shahzad a &eacute;crit&nbsp;:<br>
                </div>
              </div>
              <blockquote 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>
                    <div class="h5">
                      <div><br>
                      </div>
                      <div>Call A: PABX -&gt; FS -&gt; Carrier -&gt;
                        User A</div>
                      <div>Call B: PABX -&gt; FS -&gt; Carrier -&gt;
                        User B</div>
                      <div><br>
                      </div>
                      <div>And then bridge them on FS, such that,</div>
                      <div><br>
                      </div>
                      <div>User A &lt;- Carrier &lt;- FS -&gt; Carrier
                        -&gt; User B</div>
                      <div><br>
                      </div>
                      <div>I think there are easier ways to achieve
                        this. But anyway in this scenario,</div>
                      <div><br>
                      </div>
                      <div>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><br>
                      </div>
                      <div>
                        <pre style="line-height:1.1em;background-color:rgb(249,249,249);border:1px dashed rgb(47,111,171);padding:1em">&lt;param name="inbound-late-negotiation" value="true"/&gt;</pre>
                      </div>
                      <div><br>
                      </div>
                      <div>So, that FS does not Answer it with local
                        SDP. Additionally you may need to disable SOA,</div>
                      <div><br>
                      </div>
                      <div>
                        <pre style="line-height:1.1em;background-color:rgb(249,249,249);border:1px dashed rgb(47,111,171);padding:1em">&lt;param name="enable-soa" value="false"/&gt;</pre>
                      </div>
                      <div><br>
                      </div>
                      <div>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><br>
                      </div>
                      <div>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><br>
                      </div>
                      <div><br>
                      </div>
                      <div> Thank you.</div>
                      <div><br>
                      </div>
                      <div><br>
                      </div>
                    </div>
                  </div>
                </div>
                <div>
                  <div class="h5">
                    <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>
                      <br>
                      <br clear="all">
                      <div><br>
                      </div>
                      -- <br>
                      <div><span
style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">Mit


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


                        Shahzad</span><br
style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">
                      <span
style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">------------------------------</span><span
style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">-----</span><br
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                      <span
style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">CISCO


                        Rich Media Communication Specialist (CRMCS)</span><br
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                      <span
style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">CISCO


                        Certified Network Associate (CCNA)</span><br
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                      <span
style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">Cell:&nbsp;+49


                        176 99 83 10 85</span><br
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                      <span
style="color:rgb(136,136,136);font-size:13px;font-family:arial,sans-serif">MSN:&nbsp;</span><a
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                    <br>
                    <fieldset></fieldset>
                    <br>
                    <pre>_________________________________________________________________________
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</pre>
                  </div>
                </div>
              </blockquote>
              <br>
            </div>
            <br>
_________________________________________________________________________<br>
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            <br>
          </blockquote>
        </div>
        <br>
        <br clear="all">
        <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
style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">
        <span
style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">------------------------------</span><span
style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">-----</span><br
style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">
        <span
style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">CISCO
          Rich Media Communication Specialist (CRMCS)</span><br
style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">
        <span
style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">CISCO
          Certified Network Associate (CCNA)</span><br
style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">
        <span
style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">Cell:&nbsp;+49
          176 99 83 10 85</span><br
style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">
        <span
style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">MSN:&nbsp;</span><a
          moz-do-not-send="true" href="mailto:shari_786pk@hotmail.com"
style="color:rgb(17,85,204);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)"
          target="_blank">shari_786pk@hotmail.com</a><br
style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">
        <span
style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">Email:&nbsp;</span><a
          moz-do-not-send="true"
          href="mailto:shaheryarkh@googlemail.com"
style="color:rgb(17,85,204);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)"
          target="_blank">shaheryarkh@googlemail.com</a>
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      <br>
      <pre wrap="">_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
<a class="moz-txt-link-abbreviated" href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a>
<a class="moz-txt-link-freetext" href="http://www.freeswitchsolutions.com">http://www.freeswitchsolutions.com</a>

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
<a class="moz-txt-link-freetext" href="http://www.cudatel.com">http://www.cudatel.com</a>

Official FreeSWITCH Sites
<a class="moz-txt-link-freetext" href="http://www.freeswitch.org">http://www.freeswitch.org</a>
<a class="moz-txt-link-freetext" href="http://wiki.freeswitch.org">http://wiki.freeswitch.org</a>
<a class="moz-txt-link-freetext" href="http://www.cluecon.com">http://www.cluecon.com</a>

FreeSWITCH-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a>
<a class="moz-txt-link-freetext" href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a>
UNSUBSCRIBE:<a class="moz-txt-link-freetext" href="http://lists.freeswitch.org/mailman/options/freeswitch-users">http://lists.freeswitch.org/mailman/options/freeswitch-users</a>
<a class="moz-txt-link-freetext" href="http://www.freeswitch.org">http://www.freeswitch.org</a>
</pre>
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