[SR-Users] Kamailio + Websocket issues

[David Villasmil]

Send the invite to all asterisk, whichever has it will respond.

On Thu, 25 Mar 2021 at 01:41, Vinicius Kwiecien Ruoso <vinicius at leads2b.com>
wrote:

> Hi Segiu,
>
> Yeah, I understand that won't be the way forward, so I need another
> way for Kamailio to figure out the route to the websocket when
> Asterisk is making the call.
>
> I've shared my config file, also with some attempts commented in the
> previous emails. The alternatives were to use an alias, or an encoded
> param. I think the alias would be a better alternative, but could not
> get it right yet.
>
> Greetings,
> Vinicius
>
> On Wed, Mar 24, 2021 at 7:16 PM Sergiu Pojoga <pojogas at gmail.com> wrote:
> >
> > Vinicius,
> >
> > The obvious is that PATH is broken in Asterisk's PJSIP and they won't do
> anything about it as it's marked "minor". It's been 2 yrs now that me and
> others have reported it.
> > [3]
> https://community.asterisk.org/t/wrong-d-uri-for-invites-with-pjsip-and-path/74079
> >
> > Bottom line: there's nothing wrong with Kamailio. Start looking for a
> workaround, don't bet on Sangoma to fix it any time soon, lol.
> >
> > Regards,
> > --Sergiu
> >
> >
> >
> > On Wed, Mar 24, 2021 at 4:52 PM Vinicius Kwiecien Ruoso <
> vinicius at leads2b.com> wrote:
> >>
> >> Hi!
> >>
> >> Thanks for the fast response. Sorry about not replying to the correct
> >> email, I've just entered the list and was not getting its individual
> >> emails.
> >>
> >> > So it’s an outbound call to a webrtc registered user? If so, kamailio
> >> > should route it to wherever the called user is registered.
> >>
> >> Yes, it is a call from the backend Asterisk to a user registered via
> >> the websocket. The register is not stored in Kamailio, so it needs to
> >> use the information in the INVITE message to be able to route to the
> >> correct connection.
> >>
> >> > you'll need to share your config and logs. This should work in your
> scenario.
> >>
> >> Turns out looking closer, Asterisk is not respecting the Path protocol
> >> [1] [2]. In the INVITE sent to Kamailio, there is no information about
> >> the path in the message.
> >>
> >> To me the best second approach that should work is the "alias="
> >> information in the contact. That makes sense?
> >>
> >> I'm sharing my current config as an attachment here. I'm new to
> >> Kamailio, so I might be missing something really obvious here.
> >>
> >> [1] https://issues.asterisk.org/jira/browse/ASTERISK-28211
> >> [2] https://community.asterisk.org/t/pjsip-path-module-issues/88046/12
> >>
> >> Thanks,
> >> Vinicius
> >>
> >> On Tue, Mar 23, 2021 at 6:43 PM Vinicius Kwiecien Ruoso
> >> <vinicius at leads2b.com> wrote:
> >> >
> >> > Hi!
> >> >
> >> > I'm using Kamaio in front of multiple Asterisk instances. At this
> >> > moment it works as a SIP over Websocket proxy, with rtpengine, for
> >> > browser clients to connect to Asterisk using WebRTC. I do not use the
> >> > registration module of Kamailio, as each backend Asterisk is
> >> > independent and handles its own registrations.
> >> >
> >> > Everything works great when making calls from the browser, and the
> >> > routing is correctly executed by Kamailio based on the request SIP
> >> > domain. We have an internal routing API that it calls to discover
> >> > which backend Asterisk to route the calls.
> >> >
> >> > The issue I have is when a call initiates from that backend Asterisk,
> >> > trying to reach a contact that is connected in Kamailio via the
> >> > websocket. The Asterisk sends the message to the proxy, and Kamailio
> >> > must route it to the corresponding websocket.
> >> >
> >> > I've tried a few approaches:
> >> > - using add_contact_alias + handle_ruri_alias: I have the alias with
> >> > alias=<ip>~<port>~ws in the contact registration, but for some reason
> >> > handle_ruri_alias cannot use it
> >> > - using the Path module on Asterisk, so when registering, the path is
> >> > recorded and sent back from Asterisk, Kamailio is also not respecting
> >> > that
> >> > - Using contact_param_encode and contact_param_encode and
> >> > contact_param_decode_ruri, but the encoded sip address is always the
> >> > invalid websocket, like sip:58c0ktrg at 5hp0nn5hqqv9.invalid
> ;transport=ws
> >> >
> >> > None with success. Any hints on that can be wrong? I can share more
> >> > detailed information.
> >> >
> >> > Greetings,
> >> > Vinicius
> >> _______________________________________________
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> >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
> >
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>
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-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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