<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">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.<div class=""><br class=""><div><blockquote type="cite" class=""><div class="">On Mar 27, 2015, at 3:28 PM, Mateus Dalepiane <<a href="mailto:mdalepiane@gmail.com" class="">mdalepiane@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class=""><div class=""><div class="">Hello Brian,<br class=""><br class=""></div>Thank you for the answer. We will consider using Verto in the future.<br class=""><br class="">Right now we will have to stick with WebRTC over SIP, we are using SIP.js for that.<br class=""><br class="">I ran some more tests and once the Websocket connection drops and is re-established,<br class=""></div><div class="">even if we send a re-INVITE, FS identifies it as belonging to the old call, and<br class=""></div><div class="">responds to it, after a while FS hangs up the call reporting a NORMAL_TEMPORARY_FAILURE.<br class=""><br class=""></div><div class="">If the Websocket is not disconnected, I can see that FS sends an re-INVITE to the client after a while,<br class=""></div><div class="">so I guess that what is happening is that when FS tries to send this re-INVITE it realizes that the old connection<br class=""></div><div class="">was closed and hangs up the call.<br class=""><br class=""></div><div class="">My question now is: Why FS does not update the connection information for the call once the re-INVITE from<br class="">the new connection is received?<br class=""></div></div><div class="gmail_extra"><br class=""><div class="gmail_quote">2015-03-26 15:15 GMT-03:00 Brian West <span dir="ltr" class=""><<a href="mailto:brian@freeswitch.org" target="_blank" class="">brian@freeswitch.org</a>></span>:<br class=""><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr" class="">Have you taken a look at Verto?</div><div class="gmail_extra"><br class=""><div class="gmail_quote"><div class=""><div class="h5">On Thu, Mar 26, 2015 at 12:08 PM, Mateus Dalepiane <span dir="ltr" class=""><<a href="mailto:mdalepiane@gmail.com" target="_blank" class="">mdalepiane@gmail.com</a>></span> wrote:<br class=""></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class=""><div class="h5"><div dir="ltr" class=""><div dir="ltr" class=""><div style="line-height: 1.38; margin-top: 0pt; margin-bottom: 0pt;" class=""><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;" class="">We have the following scenario: The session is established between WebRTC and FreeSWITCH using Websockets.</span></div><br class=""><div style="line-height: 1.38; margin-top: 0pt; margin-bottom: 0pt;" class=""><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;" class="">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 class=""><div style="line-height: 1.38; margin-top: 0pt; margin-bottom: 0pt;" class=""><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;" class="">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 class=""><div style="line-height: 1.38; margin-top: 0pt; margin-bottom: 0pt;" class=""><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;" class="">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>
</div></div></blockquote></div></div></blockquote></div></div></div></blockquote></div><br class=""></div></body></html>