<div dir="ltr"><div><div>Hello Michael,<br><br></div>The SIP.sj part is working as I expect, the problem as far as I understand is that FS does not realize that the connection related to the call changed.<br><br>I must admit that I don't know how FS handles SIP over TCP, but it seems to be storing the connection that start the call. I believe it would make sense to store the connection where the last re-INVITE was received.<br></div></div><div class="gmail_extra"><br><div class="gmail_quote">2015-03-27 16:43 GMT-03:00 Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word">This is not a feature in any of the sip js stacks I know of, and I'm not quite sure how it would be implemented on top of sip. As Brian said, this is a feature in verto.<span class=""><div><br><div><blockquote type="cite"><div>On Mar 27, 2015, at 3:28 PM, Mateus Dalepiane <<a href="mailto:mdalepiane@gmail.com" target="_blank">mdalepiane@gmail.com</a>> wrote:</div><br><div><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"><<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>></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>On Thu, Mar 26, 2015 at 12:08 PM, Mateus Dalepiane <span dir="ltr"><<a href="mailto:mdalepiane@gmail.com" target="_blank">mdalepiane@gmail.com</a>></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><div dir="ltr"><div dir="ltr"><div style="line-height:1.38;margin-top:0pt;margin-bottom:0pt"><span style="font-size:15px;font-family:Arial;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></div><br><div style="line-height:1.38;margin-top:0pt;margin-bottom:0pt"><span style="font-size:15px;font-family:Arial;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></div><br><div style="line-height:1.38;margin-top:0pt;margin-bottom:0pt"><span style="font-size:15px;font-family:Arial;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></div><br><div style="line-height:1.38;margin-top:0pt;margin-bottom:0pt"><span style="font-size:15px;font-family:Arial;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></div></div></div>
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