[Freeswitch-users] Re-establish connection within a SIP session

Abdul Hakeem alhakeem at gmail.com
Fri Mar 27 23:05:46 MSK 2015


Hi Guys,
What's the best recommended client to connect to Verto ?
Cheers,
Abdul Hakeem
 
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Friday, March 27, 2015 7:43 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Re-establish connection within a SIP session
 
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.
 
On Mar 27, 2015, at 3:28 PM, Mateus Dalepiane <mdalepiane at gmail.com> wrote:
 
Hello Brian,
Thank you for the answer. We will consider using Verto in the future.

Right now we will have to stick with WebRTC over SIP, we are using SIP.js for
that.

I ran some more tests and once the Websocket connection drops and is
re-established,
even if we send a re-INVITE, FS identifies it as belonging to the old call, and
responds to it, after a while FS hangs up the call reporting a
NORMAL_TEMPORARY_FAILURE.
If the Websocket is not disconnected, I can see that FS sends an re-INVITE to
the client after a while,
so I guess that what is happening is that when FS tries to send this re-INVITE
it realizes that the old connection
was closed and hangs up the call.
My question now is: Why FS does not update the connection information for the
call once the re-INVITE from
the new connection is received?
 
2015-03-26 15:15 GMT-03:00 Brian West <brian at freeswitch.org>:
Have you taken a look at Verto?
 
On Thu, Mar 26, 2015 at 12:08 PM, Mateus Dalepiane <mdalepiane at gmail.com> wrote:
We have the following scenario: The session is established between WebRTC and
FreeSWITCH using Websockets.
 
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.
 
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?
 
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?
 
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