[SR-Users] Kamailio + Websocket issues

[Sergiu Pojoga]

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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