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

[Mateus Dalepiane]

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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>
>
>
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