I am fairly new to webRTC. The problem pertains to ICE restart. Let's say there are 2 peers connected using webRTC and one of them loses connection. Now, the peer connection will first go into "disconnected" state. And shortly after, if there is still no connection, goes into "failed" state.
Now, I understand once this failed state is reached, I have to perform an ICE restart. The problem is that even though one peer loses connection, both peers will report "failed" state and try to perform ICE restart, which I believe should be problematic. Here is a snippet of the code:
if (peer.localConnection.iceConnectionState == "failed") {
// create an offer
peer.localConnection.createOffer({
iceRestart : true
}).then(function(offer) {
peer.localConnection.setLocalDescription(offer);
// forward the offer to the signaling server
var msg = createMsg("OFFER", myId, peerId, offer);
sendToSignallingServer(msg);
}, function(error) {
//error
});
}
I understand that upon finding that there are now two offers, one of the peers should perform a "rollback" using RTCSessionDescription("rollback"). But I am confused whether this will work or not since both the peers might try to perform rollback.
How I can I make sure that only one peer performs a rollback?
One way to avoid the situation (as rollback is not widely implemented yet) is to only do the ice restart when your side of the connection sent the initial offer.
Related
I have a working WebRTC JavaScript application. Here is the problem: if during a web call there is a bad network connection, the call is stopped without WebRTC attempting to reconnect.
I would like to improve the code of my application by adding an automatic reconnection attempt system, however, in order to do so I need to understand some theory about WebRTC (and I think this can be very useful for many other developers).
Here are my questions:
Does WebRTC have a native functionality to attempt reconnection if the network is bad or should I listen for some "disconnection tigger" and call "a function" to start a new connection from zero?
If things cannot be done magically, what is/are the right "disconnection tigger/s" and "the function" from which the reconnection attempt process should restart? Is there something that can (or should) be taken from the previous connection?
I have read about an {iceRestart: true} parameter. Should this be used for the first call (then WebRTC will magically handle disconnection by attempting to reconnect) or should I use it ONLY when my code attempts to reconnect (on the 2nd, 3rd times...)?
What is the difference between connectionState "disconnected", "failed" and "closed" and does it have something to do with attempting to reconnect with bad network?
What is the best way to avoid attempting to reconnect in an infinite loop if there is no hope to reconnect (i.e: internet completely down)?
What is the difference between oniceconnectionstatechange and onconnectionstatechange? which is relevant in my case?
Thanks!
Luca
I was able to find the (JavaScript) solution through experimenting...
1) Does WebRTC have a native functionality to attempt reconnection if the network is bad or should I listen for some "disconnection tigger" and call "a function" to start a new connection from zero?
Yes, it does it by default in JavaScript, unless your code handles disconnection by proactively terminating the call through additional lines of instructions.
2) If things cannot be done magically, what is/are the right "disconnection tigger/s" and "the function" from which the reconnection attempt process should restart? Is there something that can (or should) be taken from the previous connection?
Things already happen under the hood (by magic). If the code terminates the call, it is probably because the disconnection trigger (ICE connection state = disconnected OR connection state = disconnected) triggers some additional code from the app you copy/pasted from somewhere.
3) I have read about an {iceRestart: true} parameter. Should this be used for the first call (then WebRTC will magically handle disconnection by attempting to reconnect) or should I use it ONLY when my code attempts to reconnect (on the 2nd, 3rd times...)?
Not useful in this scenario.
4) What is the difference between connectionState "disconnected", "failed" and "closed" and does it have something to do with attempting to reconnect with bad network?
You have to listen for connectionState = disconnected, the other ones don't matter for this purpose.
5) What is the best way to avoid attempting to reconnect in an infinite loop if there is no hope to reconnect (i.e: internet completely down)?
No problem, the reconnection WebRTC handles automatically will not cost anything in terms of signaling, therefore, you can try to reconnect as many times as you want, the user will eventually exit the call on his own if things are too slow.
6) What is the difference between oniceconnectionstatechange and onconnectionstatechange? which is relevant in my case?
No difference in this case, the only difference is that the ice state change is triggered right before the connection state change.
--
I hope this will be helpful to somebody!
I have two peers that want to connect to each other via WebRTC. Typically the first peer would create an offer and send it to the second via a signalling channel/server, the second peer would respond with an answer. This scenario works fine.
However, is it possible to support the case where both peers happen to try to connect to each other simultaneously both sending SDP offers to one another concurrently via the signalling server.
// Both peers do this simultaneously:
const conn = new RTCPeerConnection(null);
const sdpOffer = await conn.createOffer();
await conn.setLocalDescription(sdpOffer);
signalingService.send(peerId, sdpOffer);
// At some point in the future both peers also receive an SDP offer
// (rather than answer) from the other peer whom they sent an offer to
// via the signaling service. If this was an answer we'd call
// RTCPeerConnection.setRemoteDescription, however this doesn't work for an
// offer:
conn.setRemoteDescription(peerSDPOffer);
// In Chrome results in "DOMException: Failed to execute 'setRemoteDescription' on 'RTCPeerConnection': Failed to set remote offer sdp: Called in wrong state: kHaveLocalOffer"
I even tried to "convert" the received peer offers into answers by rewriting the SDP type from offer to answer and setup:actpass to setup:active but that doesn't seem to work, instead I just get a new exception.
So the question is, is this simultaneous connect/offer use case supported in some fashion - or should I close one side/peer RTCPeerConnection & instantiate a new one using RTCPeerConnection.createAnswer this time?
This situation is known as "signaling glare". The WebRTC API does not really define how to handle this (except for something called "rollback" but it is not implemented in any browser yet and nobody has missed it so far) so you have to avoid this situation yourself.
Simply replacing the a=setup won't work since the underlying DTLS mechanism still needs the concept of a client and a server.
The answer for how to avoid glare these days is to use the Perfect Negotiation Pattern: https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/Perfect_negotiation
However, what OP described does work with the slight modification of setting setup:active on one peer and setup:passive on the other:
https://codepen.io/evan-brass/pen/mdpgWGG?editors=0010
It might not work for audio / video connections (because those may require negotiating codecs), but I've tested it on Chrome / Firefox / Safari for DataChannel only connections.
You could choose which peer is active and which is passive using whatever system you use to determine 'politeness' in perfect negotiation. One possibility would be to compare the DTLS fingerprints and make whichever one is larger the active peer.
In WebRTC, there seems to be a very well-defined order in which things happen.
Locally I use getUserMedia to get my local stream, and save the stream to a variable. I create an RTCPeerConnection object, which I name pc, and I add the local stream to it. I add an onaddstream event handler to pc, so that I can save the remote user's stream to a variable, and eventually set it as the src attribute of an HTML element like audio. I also set onicecandidate event handler on my pc to handle ice candidates.
At this point, there is an RTCPeerConnection, but no remote user "connected yet". This is where the "offer/answer" starts. Let's say I'm using websockets for signaling and I receive an offer, which is a message called 'offer' and containing an SDP object. How do I reject it and how should this be dealt with on both endpoints?
For instance, I could send a message 'reject' that would be relayed to the other user. My RTCPeerConnection still exists, and maybe I want to be able to receive other calls. As is, I don't have to do anything to my RTCPeerConnection, correct? Does the other user, who sent the offer, have to do anything? Does he have to close that particular RTCPeerConnection? I would think not, since all he did was create an SDP object, and then outside of WebRTC, through websockets, sent the object over to the other user. He did add the offer using setLocalDescription though. When the offer is rejected, does he need to do anything about this?
When I create an offer, and send it to the other user, if I never get an answer back, can I just send an offer to a third user and then if he sends an answer I'm connected with him?
I haven't found anything about the lifecycle of an RTCPeerConnection.
Proper (spec) way to reject media
The "proper" way to "reject" offered media in an answer hasn't been implemented in any browser yet:
pc.ontrack = e => e.transceiver.stop();
Basically, the WebRTC 1.0 spec has changed quite significantly in this area. In short, a transceiver is an object combining one sender and one receiver, each sending or receiving a single track.
transceiver.stop() lets you reject a single bidirectional m-line (negotiated media) in the signaled SDP media description. E.g. you can reject parts of an offer in your answer, without rejecting the whole thing.
Today
Today, the only way to reject individual m-lines is through mangling the SDP offer/answer manually.
But it sounds like you're not actually asking about that at all. Instead, it sounds like you're asking how to bail out of incomplete signaling and roll a peer connection back to "stable" state.
Rollback to stable state
The offer/answer negotiation cycle is a state machine. The state is pc.signalingState:
You asked if one side walks away from the negotiation, does either side need to do anything before they can re-purpose their connection object for a new attempt with the same or different peer. Well, it depends.
If you've only called createOffer then no rollback of state is necessary, since createOffer is not in the above diagram.
If you've called setLocalDescription however, then you're now in "have-local-offer" state, which means you do need to somehow get back to "stable" state before you can reuse the connection.
Your options are to either finish the negotiation, delete the connection, or rollback to stable state (currently only supported in Firefox, though it is in the spec):
let pc = new RTCPeerConnection();
pc.onnegotiationneeded = async e => {
try {
await pc.setLocalDescription(await pc.createOffer());
console.log(pc.signalingState); // have-local-offer
await pc.setLocalDescription({type: "rollback"});
console.log(pc.signalingState); // stable
} catch(e) {
console.log(e);
}
}
pc.createDataChannel("dummy");
<script src="https://webrtc.github.io/adapter/adapter-latest.js"></script>
Of course, you should also let the peer know, usually through out-of-band signaling.
Why this is typically not a problem.
In typical cases, you own the JavaScript on both ends, so this doesn't come up. In other words, the desire to have a connection usually precedes one being made.
I'm using this AutobahnJS code in node to receive data from a service. It works great, getting multiple events per second. When my internet temporarily disconnects Autobahn does not detect the lost connection and does not write "Websocket connection dropped" to console, it just hangs. Indefinitely.
Is there a timeout one can set, if no data arrives after 1 minute, reconnect? Or can I use a setTimeout function to ping a server and if no pong returns close the connection and try to reopen it?
I googled till my fingers were bleeding, but I didn't find a straightforward answer to this question. Thank you very much!
connection.onopen = function(session) {
session.subscribe(arg, someEvent);
}
connection.onclose = function() {
console.log("Websocket connection dropped");
}
connection.open();
It is not possible to recognize an unclean disconnect without some data being sent. The WebSocket ping/pong mechanism at the protocol level is not exposed in the browser, and Autobahn|JS does not have any different handling when running in Node.js.
For the time being, you need to implement your own ping/pong mechanism at the application level.
I am working on webRTC video calling. I got datachannel successfully implemented. Now I would like to add video stream to the same peer connection.
I have read that stream should be added before answer and offer. Is there a way to add stream after answer or offer?
In case I have added stream before offer or answer, how could I stop streaming and start it again when needed?
Could there be any issues in maintaining so many streams?
To add stream after creating complete signalling, the Peer connection should renegotiate with stream.
pc1.addstream(stream)
Then once again create offer and send it to other Peer.
Remote peer will add stream and send answer SDP.
To stop streams:
stream.stop();
pc1.removeStream(stream);
In my experience, what Konga Raju advised didn't work. I couldn't send an "updated offer" and have the video streaming actually happen.
I found that this sequence of events works for my case, in which I wish to stream video from peer 1 to peer 2.
set up some way for the peers to exchange messages. (The variance in how people accomplish this is what makes different WebRTC code samples so incommensurable, sadly.)
On each side, set up handlers for the important signalling events. (Some folks have reported that you need to create these handlers at special times, but I haven't found that to be the case.
) There are 3 basic events:
an ice candidate sent from the other side ==> call addIceCandidate with it
an offer message ==> SetRemoteDescription & make an answer & send it
an answer message ===> SetRemoteDescription
On each side, create the peerconnection object with the event handlers we care about: onicecandidate, onremovestream, onaddstream, etc.
ice candidate pops out of the peerconnection object ===> send it to other side
When both peers are present and all the handlers are in place, peer 1 gets a trigger message of some kind to start video capture (the getUserMedia call)
Once getUserMedia succeeds, we have a stream. Call addStream on the peer connection object.
Then peer 1 makes an offer
Due to the handlers we set up earlier, peer 2 sends an answer
Concurrently with this (and rather opaquely), the peer connection object starts producing ice candidates. They get sent back and forth between the two peers and handled (steps 2 & 3 above)
Streaming starts by itself, opaquely, as a result of 2 conditions:
offer/answer exchange
ice candidates received, exchanged, and handled
I haven't found a way to add video after step 9. When I want to change something, I go back to step 3.
MediaStream should be added to peerconnection first only then exchange of offer, answer ,candidates should be done. If the onAddStream() is called ,that mean you are receiving the remote video.