[SR-Users] Terminate dialogs on [websocket:closed]
mayamatakeshi at gmail.com
Wed Feb 10 20:57:25 CET 2016
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.
On Wed, Feb 10, 2016 at 10:23 PM, Daniel-Constantin Mierla <
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
> 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,
> On Wed, Feb 10, 2016 at 5:59 PM, Daniel-Constantin Mierla <
> <miconda at gmail.com>miconda at gmail.com> wrote:
>> 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.
>> On 09/02/16 23:37, mayamatakeshi wrote:
>> 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
>> 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
> 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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