Ggtcf

`p2p:
// Enables peer to peer mode. When enabled the system will try to
// establish a direct connection when there are exactly 2 participants
// in the room. If that succeeds the conference will stop sending data
// through the JVB and use the peer to peer connection instead. When a
// 3rd participant joins the conference will be moved back to the JVB
// connection.
enabled: true,

 // Use XEP-0215 to fetch STUN and TURN servers.
 useStunTurn: true,

 // The STUN servers that will be used in the peer to peer connections
 stunServers: [
 urls: 'stun:bturn2.xirsys.com' ,
 urls:'turn:bturn2.xirsys.com:80?transport=udp',username:'xxxxxxxxxxxxxxxxxx',credential:'xxxxxxxxxxxxxxxxxxxxxxxxx',
 urls:'turn:bturn2.xirsys.com:3478?transport=udp',username:'xxxxxxxxxxxxxxxxxx',credential:'xxxxxxxxxxxxxxxxxxxxxxxxx',
 urls:'turn:bturn2.xirsys.com:80?transport=tcp',username:'xxxxxxxxxxxxxxxxxx',credential:'xxxxxxxxxxxxxxxxxxxxxxxxx',
 urls:'turn:bturn2.xirsys.com:3478?transport=tcp',username:'xxxxxxxxxxxxxxxxxx',credential:'xxxxxxxxxxxxxxxxxxxxxxxxx'
 ],

 // Sets the ICE transport policy for the p2p connection. At the time
 // of this writing the list of possible values are 'all' and 'relay',
 // but that is subject to change in the future. The enum is defined in
 // the WebRTC standard:
 // https://www.w3.org/TR/webrtc/#rtcicetransportpolicy-enum.
 // If not set, the effective value is 'all'.
 // iceTransportPolicy: 'all',

 // If set to true, it will prefer to use H.264 for P2P calls (if H.264
 // is supported).
 preferH264: true

 // If set to true, disable H.264 video codec by stripping it out of the
 // SDP.
 // disableH264: false,

 // How long we're going to wait, before going back to P2P after the 3rd
 // participant has left the conference (to filter out page reload).
 // backToP2PDelay: 5
,
useStunTurn: true, // use XEP-0215 to fetch STUN and TURN server for the JVB connection
useIPv6: false, // ipv6 support. use at your own risk
useNicks: false,`