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 < mico...@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 < > <mico...@gmail.com>mico...@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-us...@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@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 > >
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