[Freeswitch-users] Re-establish connection within a SIP session
Michael Jerris
mike at jerris.com
Fri Mar 27 23:15:01 MSK 2015
verto has its own JS client in tree.
> On Mar 27, 2015, at 4:05 PM, Abdul Hakeem <alhakeem at gmail.com> wrote:
>
> 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 <mailto: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 <mailto: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 <mailto: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 utilFreeSWITCH 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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