awesome,<div><br></div><div>I call that kind of fix a google(tm) commit.</div><div><br></div><div>Its the equiv of pressing the [ I'm feeling lucky ] button ;)</div><div><br><br><div class="gmail_quote">On Thu, Sep 27, 2012 at 2:03 AM, Peter Olsson <span dir="ltr"><<a href="mailto:peter.olsson@visionutveckling.se" target="_blank">peter.olsson@visionutveckling.se</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Tony, your change fixed the problem!<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Thank you so much :)<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt;font-family:"Calibri","sans-serif";color:#1f497d">/Peter<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">Från:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>]
<b>För </b>Anthony Minessale</span></p><div class="im"><br>
<b>Skickat:</b> den 26 september 2012 21:28<br>
<b>Till:</b> FreeSWITCH Users Help<br>
<b>Ämne:</b> Re: [Freeswitch-users] Sometimes missing a few seconds of audio when CN is offered in INVITE (and solution)<u></u><u></u></div><p></p><div class="im">
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">Looking at the code, I can see a solution assuming your theory is correct.<u></u><u></u></p>
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<p class="MsoNormal">I pushed a patch to HEAD if you care to test it.<u></u><u></u></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p>
<div><div class="im">
<p class="MsoNormal">On Wed, Sep 26, 2012 at 3:26 AM, Peter Olsson <<a href="mailto:peter.olsson@visionutveckling.se" target="_blank">peter.olsson@visionutveckling.se</a>> wrote:<u></u><u></u></p>
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<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt">Hello everyone! I experience a strange issue for bridged calls from Lync -> FreeSWITCH -> Asterisk, it seems to be related to CN packets.
I’m not sure if this a bug or not so I ask the question here first.</span><u></u><u></u></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt"> </span><u></u><u></u></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt">The problem is that the audio back from the Asterisk server is not bridged back to the originating client for the first few seconds (how
long is different on different calls), instead FS seems to send CN-packets back. I’ve looked into a wireshark dump, and I see the audio is coming from the Asterisk server (using the correct IP and port), but FS doesn’t write the same packet to the other call
leg (back to Lync), instead it sends a CN packet every second or so. My guess is that the timestamp is being handled wrong somehow, and FS believes that the real packet is too old (since it has already sent a CN packet with a higher timestamp?), and should
not be written to the other leg, then after a while it sends a new CN packet and so on.</span><u></u><u></u></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt"> </span><u></u><u></u></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt">The solution is to set “suppress_cng=true” before bridging the call to the Asterisk server, when this is done, the audio is always bridged
correctly, and nothing is missing.</span><u></u><u></u></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt"> </span><u></u><u></u></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt">I have pcaps of both working calls (with the variable set) and nonworking calls, so if you believe it might be something that FS should
handle differently I’ll submit this to Jira.</span><u></u><u></u></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt;color:#888888"> </span><span style="color:#888888"><u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt;color:#888888">/Peter</span><span style="color:#888888"><u></u><u></u></span></p>
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</div><p class="MsoNormal" style="margin-bottom:12.0pt"></p><div class="im"><br>
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<p class="MsoNormal"><u></u> <u></u></p>
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<p class="MsoNormal">-- <br></p><div class="im">
Anthony Minessale II<br>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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