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<p class="MsoNormal">Hello All,<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I am trying to resolve, in my mind, the flow of a WebRTC<span style="font-family:Wingdings">ó</span>WebRTC call using Kamailio and Asterisk.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Each WebRTC client is registered in Kamailio and when I call WebTRC Client1 from WebRTC Client2 what I see is -><o:p></o:p></p>
<p class="MsoNormal">The Invite is sent from Kamailio to Asterisk and then Asterisk is sending the Invite back to Kamailio. Also depending on<o:p></o:p></p>
<p class="MsoNormal">The version of Asterisk, the INVITE will then get forwarded to the AOR that is registered in Kamailio for the called number.<o:p></o:p></p>
<p class="MsoNormal">Does this seem correct? It seems like there is an extra hop in there.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">The reason I am now very curious now is because everything works fine if using Kamailio 5.0 and Asterisk 14.6, but I switch to Asterisk 15.3<o:p></o:p></p>
<p class="MsoNormal">I get the extra hop and call is dropped after 30 seconds.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I would appreciate any thoughts on this.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Thank you in advance.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
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