ICE Error on SipTest

General
1

Hi, everyone!

I have a strange problem with Plugin Demo Sip Gateway.

Janus and Coturn are installed on the server.
Echotest - works great and fast.

But Siptest can’t establish a connection and disconnects with error 408 - Timeout.
Same server, same settings.

More details about the problem:

In Admin/Monitor - only one “sdps” record - remote

"sdps": {
        "profile": "UDP/TLS/RTP/SAVPF",
        "remote": "v=0\r\no=- 5604222774015545125 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 ......"
    }

and in webrtc>ice section only “local-candidates” exists

"local-candidates": [
                "1 1 udp 2013266431 91.198.63.42 58573 typ host",
                "1 1 udp 2013266431 10.1.96.65 58573 typ host",
                "2 1 tcp 1015022079 91.198.63.42 0 typ host tcptype active",
                "2 1 tcp 1015022079 10.1.96.65 0 typ host tcptype active",
                "3 1 tcp 1010827775 91.198.63.42 54583 typ host tcptype passive",
                "3 1 tcp 1010827775 10.1.96.65 54583 typ host tcptype passive"
            ],

also, i`m see

"pending-trickles": 7,

and, in janus.log i have multiple

[5519818092069964] Still waiting for the answer, queueing this trickle to wait until we're done there...

In Crome - webrtc-internals i have this status:

ICE connection state: new
Connection state: new
Signaling state: new => have-local-offer
ICE Candidate pair: (not connected)
ICE candidate grid

and this icecandidateerror

url: turn:media-vb-test.bcc.kz:3478?transport=tcp
address: 192.168.0.x
port: 16823
host_candidate: 192.168.0.x:16823
error_text: Address not associated with the desired network interface.
error_code: 600

At the moment I have run out of ideas what is going wrong and why only on SipTest.
I really hope for your help.

Best Regards, Vitaliy.

2

408 is a SIP error, not a WebRTC error. Check the SIP traffic.

3

Good day, Lorenzo!
You are absolutely right.

More detailed about problem:

We are have SIP trouth Cisco PCCE.

After start call and SIP plugin send INVITE message on 5060 UDP port,
and I see voice trafic on RTP/RTCP port (39500 in my case).

But no RINGING (200) answer was recive from SIP server, and we are have timeout.

Solution was very simple.

"force_tcp" : <true|false; if true, forces TCP for the SIP messaging; optional>,

// We're registering as guests, no username/secret provided
var register = {
  request: "register",
  type: "guest",
  force_tcp: true << added in siptest.js
};

I hope this will be useful, especially for those who are first time setting up SIP.

Best regards, Vitaliy.

P.S.
I’m still trying to figure out if this is Cisco specific or NAT/Firewall is misconfigured.
Unfortunately, I don’t have access to these servers.