[SR-Users] Terminate dialogs on [websocket:closed]

[mayamatakeshi]

Hello,
I checked the docs and indeed sip.js supports GRUU (haven't heard of this
before).
I will try with it.
Thanks,
Takeshi

On Thu, Feb 11, 2016 at 6:01 AM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

> Hello,
>
> if the client UA does GRUU (iirc, jssip supports that, sip.js started as a
> fork of jssip) and you enable that in Kamailio (see registrar module), it
> should be enough that the client UA reconnects on websocket for SIP
> singaling, no need for re-INVITE, unless the IP/ports for media stream
> change. The BYE or other requests within dialog will be routed properly to
> the new contact address after the reconnect.
>
> But you have to check with your UA and see how it behaves in order to
> build the proper solution on server side.
>
> Cheers,
> Daniel
>
>
> On 10/02/16 20:57, mayamatakeshi wrote:
>
> Hi Daniel,
> originally I was thinking that i should just terminate or allow the dialog
> module to terminate the call if websocket:closed happens.
> However, I believe the sip.js library I am using in the client will do a
> re-INVITE after websocket connection is reestablished so that requests from
> the other end will come to the right socket so I am thinking in just reduce
> the dlg timeout and if the re-INVITE happens, reset it to its usual value.
> I have not checked yet if sip.js does this re-INVITE but I believe it is
> reasonable to assume it does and if it doesn't I think it should not be
> difficult to patch it to do so.
>
> I will check the other alternatives you mentioned.
> Thank you.
>
> Regards,
> Takeshi
>
> On Wed, Feb 10, 2016 at 10:23 PM, Daniel-Constantin Mierla <
> <miconda at gmail.com>miconda at gmail.com> wrote:
>
>> Hello Takeshi,
>>
>> so do you expect a re-INVITE after the websocket connection is closed?
>>
>> You may want to check also the dialog keepalive features, it might just
>> be enough to enable it, but of course it may take longer to detect when one
>> leg of the call is gone.
>>
>> Also, typically with PSTN gateways works to set session timers (see sst
>> module).
>>
>> Cheers,
>> Daniel
>>
>>
>> On 10/02/16 12:27, mayamatakeshi wrote:
>>
>> Hi Daniel,
>>
>> Yes, that will solve it.
>> Then when i get the in-dialog INVITE i can revert the lifetime back to
>> the original value.
>> Thanks and regards,
>> Takeshi
>>
>> On Wed, Feb 10, 2016 at 5:59 PM, Daniel-Constantin Mierla <
>> <miconda at gmail.com>miconda at gmail.com> wrote:
>>
>>> Hello,
>>>
>>> perhaps you can just lower the dialog lifetime in the websocket event
>>> route, then dialog will take care of sending the BYEs, without the need to
>>> store additional information in hash table.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>> On 09/02/16 23:37, mayamatakeshi wrote:
>>>
>>> Hello,
>>> I am using module websocket and it works very well.
>>> However I would like to send BYE to the other end if event
>>> [websocket:closed] happens.
>>> From the docs I can see websocket module itself doesn't provide for this.
>>>
>>> I was considering doing something like this:
>>>   - use module htable to match $si:$sp to dialogs
>>>   - use event_route[dialog:start] to insert dialog info to my htable
>>> under $si:$sp of Websocket side of the call
>>>   - use event_route[dialog:end] to remove dialog info from htable
>>>   - use event_route[websocket:closed] to iterate over entries in the
>>> htable under key $si:$sp and call dlg_get() and dlg_bye().
>>>
>>> Obs: in the above, there is a risk of losing some dialogs as insertion
>>> in htable cannot be done atomically, but I am fine with it as it it not
>>> expected to happen as WebSocket users would only infrequently generate
>>> simultaneous calls.
>>>
>>> However before going with this, I would like to ask for other possible
>>> approaches.
>>>
>>> Regards,
>>> Takeshi
>>>
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>> --
>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>> Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>> --
>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
>>
>>
>
> --
> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
>
>
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