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<p class=MsoNormal><span lang=EN-US style='color:#1F497D'>Hello all,<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='color:#1F497D'>I do a test without
RTP proxy to check if it’s the problem.<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='color:#1F497D'>To do this test I use
same cisco GW for the incoming and outgoing call, and I have exactly the same
problem.<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='color:#1F497D'>Laurent<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span lang=FR style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>De :</span></b><span
lang=FR style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Laurent
Schweizer [mailto:laurent.schweizer@peoplefone.com] <br>
<b>Envoyé :</b> samedi 25 juillet 2009 18:01<br>
<b>À :</b> cisco-voip@puck.nether.net<br>
<b>Objet :</b> [cisco-voip] Early media with cisco 5400<o:p></o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span lang=EN-US>Hello,<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US>I have a cisco 5400 and I have a problem
with early media.<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US>When I receive a call from PSTN to voip and
then this call is forwarded to a PSTN destination via an other GW, this other
GW send me back a 183 Session in progress with SDP .<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US>The problem is that the GW is not
transferring the voice to the PSTN or as I use a RTP proxy, maybe
the cisco AS5400 don’t send RTP traffic, so the RTP proxy can’t
start the relay (he wait 1 RTP packet to detect the publicIP/port)<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US>When the user answer , the tha call is OK
in both way.<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US>any idea how to solve this ?<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US>In my config I have set:<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US>voice call send-alert<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US>voice rtp send-recv<o:p></o:p></span></p>
<p class=MsoNormal>!<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>dial-peer voice 2 voip<o:p></o:p></p>
<p class=MsoNormal> destination-pattern 2T<o:p></o:p></p>
<p class=MsoNormal> <span lang=EN-US>progress_ind setup enable 3<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US> progress_ind progress enable 8<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US> translate-outgoing called 1<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US> voice-class codec 7<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US> session protocol sipv2<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US> session target ipv4:203.40.5.100<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US> dtmf-relay rtp-nte<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US> no vad<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US>!<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US>gateway<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US>!<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US>sip-ua<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US> nat symmetric check-media-src<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US> retry invite 8<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US> retry bye 2<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US> retry cancel 2<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US> timers expires 120000<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='color:#1F497D'>link to 2 calls,
first only 183 session in progress and secondl, an answered calls:<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='color:#1F497D'><a
href="http://pastebin.com/m26a0e423">http://pastebin.com/m26a0e423</a><o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='color:#1F497D'><o:p> </o:p></span></p>
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