WebRTC ICE fails to connect when both users set ICE transportpolicy to relay
I am currently working on p2p video calling app and forcing iceTransportPolicy to relay for both the users for testing. I am not able to view streams on the both the users end in this case because the ICE connectivity just fails when both the user's ICE policy are set to relay. But when I am using iceTransportPolicy as relay only for one user(either of them) and other user uses srflx/prflx candidates, the TURN server works just fine and I am able to view streams at both the users end.
I am using a coturn TURN server hosted in AWS with all necessary ports open for it to work successfully. The turn server is also configured properly(according to my understanding) as I can see all kind of candidates are being generated while testing with trickle ICE test page. Here is the test result from trickle ICE.
Can there be an explanation for such a behavior?
As mentioned in the above comment, the TURN createPermission request was failing with error code 403. The reason was a wrong configuration at the turnserver.conf file.
As mentioned in the default turnserver.conf file, for Amazon Ec2 users, it is important to understand the below line.
TURN Server public/private address mapping, if the server is behind NAT.
As my TURN server was indeed hosted in aws ec2, I needed to do a public / private mapping which I was not doing.
My external-ip line in turnserver.conf looked like
external-ip = 1.2.3.4 where as it should be external-ip = 1.2.3.4/172.x.x.x
After making the changes and restarting the turn server, now my service works like a charm.
Keep in mind that this is only needed if your turn server is behind a NAT which is the case with a cloud provider like AWS.
Hope it helps a future reader facing similar issues.
- Change playout delay in WebRTC stream
- Setting resolution for WebRTC using Unity Render Streaming to stream video file in a specific resolution
- One-Way Audio on 1:1 call .NET MAUI , Azure Communication Services using Javascript
- How to check Microphone Volume in Angular
- WebRTC never fires onIceCandidate
- Flutter receive web rtc calls when app is terminated (detached state)
- WebRTC Data channel is opened but not sending any data
- WebRTC - multi-track streams issue on Firefox <> Chrome
- getUserMedia how can i improve audio quality? How can I make audio to stop echoing?
- Generate ECDSA BouncyCastle certificate without EC named parameters
- Send text message from Unity Client to Javascript server using Unity Render Streaming and WebRTC
- Can I use PeerJS without a PeerJS Server?
- Undefined reference to 'owr_init' (linker error)
- Cannot figure out why DTLS handshake does not complete
- android webrtc intent leak
- webRTC - Failed to set remote answer only for public (LAN or same access point works fine)
- Coturn stun server returned an error with code=701
- android webRTC video not woroking
- getUserMedia() and webRTC, modify the buffer before sending
- How to make media recorder api individual chunks playable by it self
- WebRTC stream not found in track when trying to screen share android device screen
- How can I authenticate PeerJS connections to the PeerServer?
- Streaming RTSP Locally through HTML5
- How to implement Low Latency, 60 fps video encoding in WebRTC in C# with WebRTC?
- Avatar failed to start. Error: InvalidCharacterError: Failed to execute 'atob' on 'Window': The string to be decoded is not correctly encoded
- When does DTLS negotiation start in WebRTC?
- Why connectionState property of RTCPeerConnection object is not accesssible?
- Chrome not detecting STUN success responses even if I can see the response sent via wireshark
- RTCPeerConnection is not defined
- What is saslprep ? why its required a stun username password to be used with saslprep?