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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