Janus demo test sends call directly to pbx

General
1

Hello,
I have installed janus server in one of my server in my local network
I have downloaded the siptest.html, siptest.js, settings.js and janus.js in my local computer and connected websocket to janus server in my local network. Everything works fine for registration and my sip account is registered with kamailio server through janus server. All three devices are in same LAN.In kamailio the contact address is shown as Janus server ip.
But the issue is -
When I am making call from my local computer using siptest.html, the call is directly going to kamailio server from my local computer and because of that reason the call is getting rejected by kamailio. The call should actualy go via websocket connection though janus server.
kindly help me for this issue.
I also tried with janus-gateway library in angular and that also behaved same. Kindly suggest how can i solve this, I am using janus first time.

logs from my browser console is as follows, when i tried from angular janus-gateway client

Angular is running in development mode. Call enableProdMode() to enable production mode.
janus.component.ts:24 Initializing library
janus.component.ts:33 Library initialized: true
janus.component.ts:33 Using WebSockets to contact Janus: ws://10.13.1.137:8188/janus
janus.js:793 Created session: 734999196496576
janus.js:995 Created handle: 4906770995621481
janus.component.ts:43 Plugin attached! (janus.plugin.sip, id=4906770995621481)
janus.component.ts:53 this.sipcall undefined
janus.component.ts:103 this.sipcall {session: Janus, plugin: ‘janus.plugin.sip’, id: 4906770995621481, token: null, detached: false, …}
janus.js:1626 Creating PeerConnection
janus.js:1635 Preparing local SDP and gathering candidates (trickle=true)
janus.js:1683 Adding local stream
janus.js:1685 Adding local track: MediaStreamTrack {kind: ‘audio’, id: ‘ac98f7b8-8977-4473-8b07-42ede7fc4230’, label: ‘Default’, enabled: true, muted: false, …}
janus.js:2296 Creating offer (iceDone=false)
janus.js:2406 Setting local description
janus.js:2428 Offer ready

janus server logs are as follows
Apr 14 16:53:37 ubuntu janus[1894]: [6715587489114799] WebRTC resources freed; 0x7fddec002090 0x7fdda800c620
Apr 14 17:06:07 ubuntu janus[1894]: Destroying session 1758689402818897; 0x7fdda800c620
Apr 14 17:06:08 ubuntu janus[1894]: Detaching handle from JANUS SIP plugin; 0x7fddec002090 0x7fdda4012640 0x7fddec002090 0x7fddec006d90
Apr 14 17:06:08 ubuntu janus[1894]: [WSS-0x7fdda40018a0] Destroying WebSocket client
Apr 14 17:06:08 ubuntu janus[1894]: [6715587489114799] Handle and related resources freed; 0x7fddec002090 0x7fdda800c620
Apr 14 17:06:08 ubuntu janus[1894]: Creating new session: 734999196496576; 0x7fddec001bd0
Apr 14 17:06:08 ubuntu janus[1894]: Creating new handle in session 734999196496576: 4906770995621481; 0x7fddec001bd0 0x7fddec002090
Apr 14 17:06:15 ubuntu janus[1894]: [4906770995621481] Creating ICE agent (ICE Full mode, controlled)
Apr 14 17:06:47 ubuntu janus[1894]: [janus.plugin.sip-0x7fdda400f180] No WebRTC media anymore
Apr 14 17:06:47 ubuntu janus[1894]: [4906770995621481] WebRTC resources freed; 0x7fddec002090 0x7fddec001bd0

2

I’m not sure I understand your problem. When you use the SIP plugin, Janus will originate SIP dialogs on behalf of users, and so when talking to Kamailio it’s normal for SIP over UDP between Janus and Kamailio. Janus doesn’t support SIP over WebSockets.

If you need SIP over WebSockets, you may want to handle signalling yourself, and only involve Janus for the media, using for instance the NoSIP plugin. You can check this presentation for an overview of the different options you have available (it was done at the OpenSIP Summit but the same exact considerations apply to Kamailio and other SIP servers as well).

3

Thank you Lorenzo, I will explain
I have connected to janus through websocket (ws://janusip:8018/janus) from my computer using janus library. I want janus to send invite to kamailio using UDP. my computer, janus and kamailio all are in LAN . But the call is not reaching janus server.
I took the tcpdump from kamailio and found that the invite is orginating from my computer to kamailio directly, in via header I can see janus ip, but in recieved from I am getting client IP. Wireshark saying the call is orginated from client ip and not from janus.
what may be the issue?, I have tried with janus sip gateway demo available in the internet and it works fine. But the issue is only when I tried with the janus I have installed in my server. Please note that I have downloaded siptest.html, siptest.js, settings.js, janus.js in my local computer and changed server variable to remote janus server.
What may be the mistake I am doing?