[SR-Users] Re: using kamailio to send invite to webrtc clients (with freeswitch)

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 &lt;shwetank.singh1(a)gmail.com&gt; Reply-To: Kamailio (SER) - Users Mailing List &lt;sr-users(a)lists.kamailio.org To: sr-users(a)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: 1004(a)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 __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: