it’s been quite a long time since I’ve posted here. Just wanted to bounce this off of you as it has been making my brain hurt. So, I have been developing a real time video chat app with WebRTC. Now, I know that the obligatory “it’s somewhere in the network stack (NAT)” answer always applies.
As is always the case it seems with WebRTC, it works perfectly in my browser and on my laptop between tabs or between Safari/Chrome. However, over the internet on HTTPS on a site I’ve created, it is shotty at best. It can accept and display the media stream from my iPhone but it cannot receive the media stream from my laptop. It just shows a black square on the iPhone for the remote video.
Any pointers would be most appreciate though as I’ve been going crazy. I know that TURN servers are an inevitable aspect of WebRTC but I’m trying to avoid employing that.
So, here is my Session class which handles essentially all the WebRTC related client side session logic:
(The publish
method is just an inherited member that emulates EventTarget/EventEmitter functionality and the p2p config is just for Google’s public STUN servers)
class Session extends Notifier {
constructor(app) {
super()
this.app = app
this.client = this.app.client
this.clientSocket = this.client.socket
this.p2p = new RTCPeerConnection(this.app.config.p2pConfig)
this.closed = false
this.initialize()
}
log(message) {
if (this.closed) return
console.log(`[${Date.now()}] {Session} ${message}`)
}
logEvent(event, message) {
let output = event
if (message) output += `: ${message}`
this.log(output)
}
signal(family, data) {
if (this.closed) return
if (! data) return
let msg = {}
msg[family] = data
this.clientSocket.emit("signal", msg)
}
initialize() {
this.p2p.addEventListener("track", async event => {
if (this.closed) return
try {
const [remoteStream] = event.streams
this.app.mediaManager.remoteVideoElement.srcObject = remoteStream
} catch (e) {
this.logEvent("Failed adding track", `${e}`)
this.close()
}
})
this.p2p.addEventListener("icecandidate", event => {
if (this.closed) return
if (! event.candidate) return
this.signal("candidate", event.candidate)
this.logEvent("Candidate", "Sent")
})
this.p2p.addEventListener("connectionstatechange", event => {
if (this.closed) return
switch (this.p2p.connectionState) {
case "connected":
this.publish("opened")
this.logEvent("Opened")
break
// A fail safe to ensure that faulty connections
// are terminated abruptly
case "disconnected":
case "closed":
case "failed":
this.close()
break
default:
break
}
})
this.clientSocket.on("initiate", async () => {
if (this.closed) return
try {
const offer = await this.p2p.createOffer()
await this.p2p.setLocalDescription(offer)
this.signal("offer", offer)
this.logEvent("Offer", "Sent")
} catch (e) {
this.logEvent("Uninitiated", `${e}`)
this.close()
}
})
this.clientSocket.on("signal", async data => {
if (this.closed) return
try {
if (data.offer) {
this.p2p.setRemoteDescription(new RTCSessionDescription(data.offer))
this.logEvent("Offer", "Received")
const answer = await this.p2p.createAnswer()
await this.p2p.setLocalDescription(answer)
this.signal("answer", answer)
this.logEvent("Answer", "Sent")
}
if (data.answer) {
const remoteDescription = new RTCSessionDescription(data.answer)
await this.p2p.setRemoteDescription(remoteDescription)
this.logEvent("Answer", "Received")
}
if (data.candidate) {
try {
await this.p2p.addIceCandidate(data.candidate)
this.logEvent("Candidate", "Added")
} catch (e) {
this.logEvent("Candidate", `Failed => ${e}`)
}
}
} catch (e) {
this.logEvent("Signal Failed", `${e}`)
this.close()
}
})
this.app.mediaManager.localStream.getTracks().forEach(track => {
this.p2p.addTrack(track, this.app.mediaManager.localStream)
})
}
close() {
if (this.closed) return
this.p2p.close()
this.app.client.unmatch()
this.logEvent("Closed")
this.closed = true
}
}