I am new to WebRTC.
I was learned about the turn server.
The below thing is used to configure the turn server works on TCP for a webrtc application.
Example turn server configuration in webrtc application :-
{
url: ‘turn:192.158.29.39:3478?transport=tcp’,
credential: ‘JZEOEt2V3Qb0y27GRntt2u2PAYA=’,
username: ‘28224511:1379330808′
}
The meaning for the ?transport=tcp is it works on TCP protocol like that they said.
My question is,
The turn server works on TCP means, it is only for establishing a connection between turn server it works on TCP or for transferring a WebRTC media via turn server it uses the TCP protocol. For which purpose the TCP packet is used ?
Is it possible to transfer the WebRTC media via the TCP packet ?
Forcing WebRTC media over TCP is not recommended, its just a fallback option for strict firewalls.
If you specify ?transport=tcp in TURN urls, then WebRTC client will connect to TURN Server over TCP.
Clients will send STUN requests over TCP to allocate relay candidates (As far as i know, currently chrome/firefox only request UDP relay candidates).
To force strict TCP via TURN server:
- Use only TURN url with
?transport=tcp
- Specify
iceTransportPolicy:"relay", so that all media will flow via TURN
After exchanging the relay candidates(udp) between peerConnections, the media path will be as below
P1 <--TCP--> P1TURN <--UDP--> P2TURN <--TCP--> P2
If both P1TURN & P2TURN are always same, then you can choose TURN relay path as LAN/lo interface i.e. you will get UDP relay candidates with LAN/lo IP.
If you want to avoid UDP & two times TURN in the media path, you need to use some SFU/SVC like Jitsi(supports pseudo-SSL candidates)/Janus.
Then media path will be as below
P1 <--TCP--> SFU/MCU <--TCP--> P2
