[SR-Users] Jumping on new TCP connection during a SIP session?

[Amar Tinawi]

Could this module work in scenario like in case if the call is established
and running,
And one of the UA'a lost its connection and then reconnect and send a new
register
Could the call be continued ?
Without sending new INVITE.. just like the whatsapp

On Mon, Jun 4, 2018, 2:47 PM Alfonso Pinto <alfonso.pinto at gmail.com> wrote:

> Not sure if I understood correctly your problem, but that sounds a lot
> like forking the call as soon as another REGISTER comes in for the same AoR.
> In that case, it may be worth to have a look into TSILO module, it may
> help you in your case.
>
> On Sun, Jun 3, 2018 at 1:55 PM, Abdoul Osséni <abdoul.osseni at gmail.com>
> wrote:
>
>> Hello group,
>>
>> The call flow is:
>>
>> useragent (TLS) -> Kamailio -> Sems Server.
>>
>> During the establishment of a call (before the 200OK or during a
>> retransmission of the 200OK issued by Sems Server via Kamailio),
>> if a user agent (TCP) loses its connection (the user agent is on a mobile
>> network for example or the network is not reliable.) and sends a
>> "register" to Kamailio (same source IP address but a different source
>> TCP port), is it possible for Kamailio to take into account this new TCP
>> port with the function t_relay () ?
>>
>> Abdoul
>>
>>
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>>
>>
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