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

[Mateus Dalepiane]

Hello Michael,

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.

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.

2015-03-27 16:43 GMT-03:00 Michael Jerris <mike at jerris.com>:

> 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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