10 Feb
2016
10 Feb
'16
9:57 p.m.
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(a)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@gmail.com>miconda@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
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--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
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_______________________________________________
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--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu