Hi,

as Dilyan mention start small and test and then grow if needed in your
config.

The loop over is for example not needed if you switch on parallel forking.
Then you have new issues with correct branch settings to rtpengine.

And a good idea for webrtc is havfos example

https://github.com/havfo/WEBRTC-to-SIP

Kind Regards
Karsten Horsmann

Дилян Палаузов <dpa-kamai...@bapha.be> schrieb am Di., 9. Mai 2023, 22:41:

>
> Hello,
>
> I suggest to call from WebRTC a non-WebRTC client, talk and then the
> non-WebRTC client hangs up.  Check if the WebRTC client gets the BYE.
>
> Once this simple scenario works, you can try the more complicated scenario
> explained by you.
>
> In my experiments the BYE was not received with topoh.  Without topoh it
> works.
>
>
> Greetings
>   Dilyan
> -----Original Message-----
> From: shwetank singh <shwetank.sin...@gmail.com>
> Reply-To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org
> >
> To: sr-users@lists.kamailio.org
> Subject: [SR-Users] using kamailio to send invite to webrtc clients (with
> freeswitch)
> Date: 05/09/2023 07:22:40 PM
>
> Hello experts
>
> I have a setup where webrtc client registrations (from behind a NAT / VPN)
> are on kamailio and the INVITE to begin a call is sent by freeswitch to
> kamailio which is then expected to relay it to the subscribed webrtc
> client.
>
> i understand, from documentation and general thinking, that kamailio
> cannot open a new connection to such a client and should reuse an open
> connection
> (a websocket in my case).
>
> i have setup kamailio as generic as it may sound and expect t_relay() to
> do the magic of forwarding the invite, however, from logs i don't see
> anything indicating kamailio is using any existing connection and hence my
> outgoing invite requests are stuck showing pointed towards a vpn gateway
> which was the registered "location" of client (which is the public ip of
> VPN gateway)
>
> please advise what relevant information can i produce here to make the
> query more readable and clear.
>
> headed here after trying just about all the resources available on the
> topic.
>
> e.g. if my location record looks like
>
>           AoR: 1...@mydomain.com
>           Contacts:
>           - Contact:
>               Address: sip:1004@11.22.33.44:49179
>               CFlags: 64
>               CSeq: 3087
>               Call-ID: 189jss21e24ntve1ggsuli
>               Expires: 593
>               Flags: 0
>               Instance: <urn:uuid:8a936bbc-514f-4755-bed0-edc644dd4453>
>               KA-Roundtrip: 0
>               Keepalive: 1
>               Last-Keepalive: 1683652755
>               Last-Modified: 1683652755
>               Methods: 7071
>               Path: <sip:10.10.6.196:3040;received=sip:mydomain.com
> ;lr;nat=yes>
>               Q: -1
>               Received: sip:11.22.33.44:49179;transport=ws
>               Reg-Id: 1
>               Ruid: uloc-645601c4-131220-3764
>               Server-Id: 0
>               Socket: tls:1.2.3.4:4443
>               State: CS_SYNC
>               Tcpconn-Id: 3916
>               User-Agent: JsSIP 3.9.0
>
> i try to lookup the location and find the address, and then route to that
> address by setting $ru
> in sngrep it does show that kamailio is trying to send INVITE to
> 11.22.33.44:port (which is the VPN gateway) but it isn't exactly clear
> from logs if
> this is being done over an existing connection.
>
> how does kamailio identify if a connection exists for a user (for each
> contact in AOR, if multiple contacts exist for a user -- i.e. different
> browser
> sessions?)
>
>
>  if(reg_fetch_contacts("location", "$ru", "caller")) {
>     $var(i) = 0;
>     while($var(i) < $(ulc(caller=>count)))
>     {
>         $ru = $(ulc(caller=>addr)[$var(i)]);
>         $var(i) = $var(i) + 1;
>         setflag(FLT_DLGINFO);
>         dlg_manage();
>         t_relay();
>     }
>   }
>
> --
> Shwetank
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