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(a)gmail.com

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&gt; wrote:

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 &lt;miconda(a)gmail.com > 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@gmail.com>miconda@gmail.com> >> 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>> >>> 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@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(a)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 > >