Is it possible to configure a TURN server with gateway_config.json?

[sooyon_l_]

Hi,

I'm testing SIP Gateway(v3.3).

I've set the TURN server info to gateway_config.json as below. but it seemed not to be applied.

-----------------------------------------------------------

I replaced my real ip/name/password.... 

{
  "gateway": {
    "_comment": "Gateway signaling server configuration",
    "publicIP": "",
    "port": 8080,
    "sport": 8443,
    "defaultVideoBW": 1024,
    "maxVideoBW": 2048,
    "stunServerUrl": "",
    "turnServer": {
      "url": "MyTurnServer.com",
      "username": "MyName",
      "password": "MyPass"
    },
   .......................

As I thought that SIP Gateway made a SDP with my turn server info. but it still had a SIP gateway ip address in SDP.

Anyone who know how I can set the turn server to sip gateway?

Thanks in advance. 

[Lei_Z_Intel1]

Sip gateway server doesn't support Turn, that's TURN setting is for client side. For client side TURN setting, you can also use 

array Woogeen.SipClient.setIceServers ( string/object/array  servers )

inline

This function establishes a connection to server and joins a certain conference.

Remarks:
This method accepts array (multiple ones) type of ice server item as argument. Typical description of each valid value should be as below:

• For turn: {urls: array or single "url", username: "username", credential: "password"}.
• For stun: {urls: array or single "url"}.

Each time this method is called, previous saved value would be discarded. Specifically, if parameter servers is not provided, the result would be an empty array, meaning any predefined servers are discarded.

Parameters

servers

turn or stun server configuration.

Returns

Result of the user-set of ice servers. 
 

<script type="text/JavaScript">

...

client.setIceServers([{

urls: "stun:x.x.x.x:3478"

}, {

urls: ["turn:x.x.x.x:443?transport=udp", "turn:x.x.x.x:443?transport=tcp"],

username: "abc",

credential: "xyz"

}]);

</script>

[Jaiswal__Avinash]

Hi ,

I am using ICS WebRtc android sdk latest version 4.1 for audio and video conference call. Does i need to set Turn Server url , username and password or it is already handled by ICS WebRtc Server. If it configured at client side  then how i can implement it because i did not find any method in ConferenceClientConfiguration to set ice servers.
Thanks in advance.

Regards,

Avinash Jaiswal

[Jaiswal__Avinash]

Jaiswal, Avinash wrote:

Hi ,

I am using ICS WebRtc android sdk latest version 4.1 for audio and video conference call. Does i need to set Turn Server url , username and password or it is already handled by ICS WebRtc Server. If it configured at client side  then how i can implement it because i did not find any method in ConferenceClientConfiguration to set ice servers.
Thanks in advance.

Regards,

Avinash Jaiswal

I got the solution . We need TURN server in case of public network for the NAT Traversal . In ICS Android sdk we can add turn server using PeerConnection.IceServer builder object and then pass it to PeerConnection.RTCConfiguration constructor and then set it to ConferenceClientConfiguration object.
Thanks,

Avinash Jaiswal