How do Stun/Turn Server + WebRTC work together?

[GenesysNewbie]

Hi Guyz,

Wondering somebody can explain how this components work together  ...

My scenario is like the following:
====================
1. A customer opens web page at home (https://800 . 148. y.z:8086/webRTC) and tries to connect/register but he cannot.
  [200.148.y.z is WebRTC Server]

2. In my understading what is happening here is that the customer is trying to reach SIP SERVER using the stun parameter
    = 800.1. a.b:3487 who is supposed to direct the call to reach SIPServer and register the client in a Dn.

3. Result, Dn is not register ... and cannot complete the process ...

How to acomplish a troubleshooting here?

Genesys does not offer good information abt STUN. How can i validate this Stun has correct parameters to "direct" calls to SIPServer.


Hope, someone can help.

thanks is advance



I cann see this log in console log of web client:

tc: 182.797: Requested access to local media with mediaConstraints: {"audio":true,"video":{"mandatory":{"maxWidth":320,"maxHeight":180}}}
grtc.js:84 chrome: {"audio":true,"video":{"mandatory":{"maxWidth":320,"maxHeight":180}}}
grtc.js:955 grtc: 182.822: User has granted access to local media
grtc.js:955 grtc: 184.462: Signing out of the WebRTC Gateway
grtc.js:955 grtc: 184.463: Signing in: https://150.20.x.y:7777/sign_in?8008
jquery-1.10.2.js:8706 GET https://150.20.x.y:9999/sign_in?8001 net::ERR_CONNECTION_TIMED_OUTsend @ jquery-1.10.2.js:8706ajax @ jquery-1.10.2.js:8136jQuery.(anonymous function) @ jquery-1.10.2.js:8282doConnect @ grtc.js:1267gClient.register @ grtc.js:1410testRegister @ demo1.js:447onclick @ audio_video.html:158
grtc.js:955 grtc: 220.531: CONNECTION_ERROR: registration failed

[jamesmurphyau]

STUN and TURN are quite different.. one is a method of discovering what IP address should be used to access a server/client (the address on the other side of the NAT) (STUN) and the other is a media relay server which is used when determining that IP address is difficult/not possible (TURN)

They are both open standards so you might be able to get some help from other areas of the internet if Genesys aren't helpful..

There appears to be some utilities that can test a STUN/TURN setup..

https://www.twilio.com/docs/api/stun-turn/faq
https://gist.github.com/roxlu/39f6cfbfd2f2e5643eac
http://stackoverflow.com/questions/28772212/stun-turn-server-connectivity-test

Do a google search you might find what you're looking for..

From the looks of it you are trying to figure out if 800.1.a.b::3487 is the publicly accessible IP/port for your SIP server?

[GenesysNewbie]

I have research lots of web pages vbut still stucked in this implementation

This STUN is given a hard time to me


I have the process running but dont know hot to test it ....

[jamesmurphyau]

Have you tried this URL:

https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/

It's in the 3rd link I pasted above.. it looks like that might help test it?

[GenesysNewbie]

Hi All

Thanks for helping, with usefull information. Good to know lots of people like to share knowledge !

BTW ... i did the install and its working, now to face video and net issues.

Have u ever try vp8?


Thanks,
GN