<div dir="ltr"><div><div>Hello Brian,<br><br></div>Thank you for the answer. We will consider using Verto in the future.<br><br>Right now we will have to stick with WebRTC over SIP, we are using SIP.js for that.<br><br>I ran some more tests and once the Websocket connection drops and is re-established,<br></div><div>even if we send a re-INVITE, FS identifies it as belonging to the old call, and<br></div><div>responds to it, after a while FS hangs up the call reporting a NORMAL_TEMPORARY_FAILURE.<br><br></div><div>If the Websocket is not disconnected, I can see that FS sends an re-INVITE to the client after a while,<br></div><div>so I guess that what is happening is that when FS tries to send this re-INVITE it realizes that the old connection<br></div><div>was closed and hangs up the call.<br><br></div><div>My question now is: Why FS does not update the connection information for the call once the re-INVITE from<br>the new connection is received?<br></div></div><div class="gmail_extra"><br><div class="gmail_quote">2015-03-26 15:15 GMT-03:00 Brian West <span dir="ltr">&lt;<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>&gt;</span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Have you taken a look at Verto?</div><div class="gmail_extra"><br><div class="gmail_quote"><div><div class="h5">On Thu, Mar 26, 2015 at 12:08 PM, Mateus Dalepiane <span dir="ltr">&lt;<a href="mailto:mdalepiane@gmail.com" target="_blank">mdalepiane@gmail.com</a>&gt;</span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5"><div dir="ltr"><div dir="ltr"><p dir="ltr" style="line-height:1.38;margin-top:0pt;margin-bottom:0pt"><span style="font-size:15px;font-family:Arial;color:rgb(0,0,0);background-color:transparent;font-weight:normal;font-style:normal;font-variant:normal;text-decoration:none;vertical-align:baseline">We have the following scenario: The session is established between WebRTC and FreeSWITCH using Websockets.</span></p><br><p dir="ltr" style="line-height:1.38;margin-top:0pt;margin-bottom:0pt"><span style="font-size:15px;font-family:Arial;color:rgb(0,0,0);background-color:transparent;font-weight:normal;font-style:normal;font-variant:normal;text-decoration:none;vertical-align:baseline">Once
 the session is established, if the websocket connection drops the media
 continues to flow util FreeSWITCH tries to send a re-INVITE to the 
client. At this point it realizes that the connection was closed and 
hangs up the call.</span></p><br><p dir="ltr" style="line-height:1.38;margin-top:0pt;margin-bottom:0pt"><span style="font-size:15px;font-family:Arial;color:rgb(0,0,0);background-color:transparent;font-weight:normal;font-style:normal;font-variant:normal;text-decoration:none;vertical-align:baseline">Now,
 if the websocket connection drops and is re-established, would it be 
possible to inform FreeSWITCH that the new connection should be used for
 the previously established session?</span></p><br><p dir="ltr" style="line-height:1.38;margin-top:0pt;margin-bottom:0pt"><span style="font-size:15px;font-family:Arial;color:rgb(0,0,0);background-color:transparent;font-weight:normal;font-style:normal;font-variant:normal;text-decoration:none;vertical-align:baseline">If
 the WebRTC client sends an INVITE message with the old session 
parameters, FreeSWITCH will be able to understand that it belongs to the
 old session?</span></p></div></div>
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<p><font face="courier new, monospace"><b><i><font size="4">Brian West</font></i></b><br><span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
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