I wonder, if it is possible to use data connections mode to make whiteboard?
Anyone attempted that? (with success)
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We want to track each users turn usage seperately. I inspected Turn Rest API, AFAIU it is just used to authorize the user which already exists in Coturn db. This is the point I couldn't realize exactly. I can use an ice server list which includes different username-credential peers. But I must have initialized these username-credential peers on coturn db before. Am I right? If I am right, I do not have an idea how to do this. Detection of user's request to use turn from frontend -> Should generate credentials like this CoTURN: How to use TURN REST API? which I am already achieved -> if this is a new user, my backend should go to my EC2 instance and run "turnadmin create user command" somehow -> then I will let the WebRTC connection - > then track the usage of specific user and send it back to my backend somehow.
Is this a true scenario? If not, how should it be? Is there another way to manage multiple users and their data usage? Any help would be appreciated.
AFAIU to get the stats data, we must use redis database. I tried to used it, I displayed the traffic data (with psubscribe turn/realm//user//allocation//traffic ) but the other subscribe events have never worked ( psubscribe turn/realm//user//allocation//traffic/peer or psubscribe turn/realm//user//allocation/*/total_traffic even if the allocation is deleted). So, I tried to get past traffic data from redis db but I couldn't find how. At redis, KEYS * command gives just "status" events.
Even if I get these traffic data, I couldn't realize how to use it with multi users. Currently in our project we have one user(in terms of coturn) and other users are using turn over this user.
BTW we tried to track the usage where we created peer connection object from RTCPeerConnection interface. I realized that incoming bytes are lower than the redis output. So I think there is a loss and I think I should calculate it from turn side.
I'm working on a WebRTC solution for audio/picture comunication and I'm a bit concerned about the lack of bandwidth control when two Peers in LAN are communicating.
Basically I want to be able to prioritize and pre-allocate bandwidth on my switch for WebRTC calls. But I couldn't see a proper way of filtering the packets when they are in a P2P call.
Also, I don't want to decode the packet to do that, because of the possible delay caused by this operation.
I hope you guys can show me a proper way or just tell me if my teorical solution could work.
I'm not 100% sure about the idea I'm planning to test, because I don't know how TURN server works internally.
But here is the idea:
And what I dont know is: Is it possible to make 2 turn servers know each other? Would they work like a 2 layer proxy between callers? If yes, could you please show me what I have to do to make it work?
Just install a internal proxy server to the external turn server and prioritize the proxy on your lan.
(answering my own question after realizing the solution was easier than I thought)
I am building a WebRTC app where two users are selected at random and then connect to each other to chat. Both clients keep an open WebSocket connection and I am planning to use this to exchange their offers/answers to signal a connection. The case I am trying to account for is when there is a peer that intentionally sends bad configuration information, and also when the peer might spontaneously disconnect in the middle of the signaling exchange.
My solution to the first case is have the server keep state of the exchange, so when the connection is first established I would expect that user A provide an offer and user B have an answer. Is this appropriate? or should this be implemented exclusively client side?
My solution to the second problem feels to me like a hack. What I am trying to do is notify the user that a match has been made and then the user will set a timeout say 20 seconds, if a connection hasn't been made in that amount of time then it should move on...
Are these appropriate solutions? How do you reliably establish a WebRTC when either peer can't be trusted? Should the signaling server be concerned with the state of the exchange?
Sounds like you're more concerned about call set up errors rather than being able to trust the identity of the remote peer. They are two very different problems.
Assuming it is the call set up errors you are concerned about you shouldn't be trying to avoid them you should be trying to make sure your application can handle them. Network connection issues are something that will always crop up and need to be handled.
Setting a timer for the establishment of a WebRTC call to complete is a logical solution. Displaying a warning to the user that the time limit is approaching also seems like a good idea. SIP is a signalling protocol and it has a defined timeout for the completion of a transaction and if it doesn't complete within that time it will generate an error response. You could use the same approach.
Scenario:
I'm using WebRTC (Google's libjingle) on iOS and PeerConnection is setup using a TURN server and I'm waiting for all candidates to gather before I send them to the peer (I'm using SIP). The problem is that although all candidates are gathered in around 1-3 seconds (I can see it in the logs) the iceGatheringChanged() callback is not called with state GatheringComplete until after around a whole minute!
Any idea why that happens?
After analyzing the traffic using Google's AppRTCDemo for iOS it seems that for GatheringComplete to fire, the client needs to already have received the candidates from the remote side, and that because it seems to need to setup TURN Allocations and add Permissions on the new allocation so that data can be exchanged with the peer. Is that the case? If so why?
Best regards
Are you exchanging the candidates for both party in real time? You are right, TURN client requires the other party candidates to create permission in TURN server and also to make check lists to start ICE processing.
I am trying to create an application which requires a user to send his local video stream to multiple peers using webRTC. As far as I've seen I am responsible for managing several PeerConnection objects because a PeerConnection can only connect to a single peer at a time. What I want to know is if it is possible to create a connection and send my local stream to a peer without him sendig his local stream to me using webRTC.
Simply don't call peer.addStream for broadcast-viewers to make it oneway streaming!
You can disable audio/video media lines in the session description by setting OfferToReceiveAudio and OfferToReceiveVideo to false.
3-Way handshake isn't drafted by RTCWebb IETF WG, yet.
Because browser needs to take care of a lot stuff simultaneously like multi-tracks and multi-media lines; where each m-line should point out a unique peer.