I want to create an "one to many" (with the max of 3 devices) webrtc setup. I have one device that is my main device. Other devices are connecting to that device. You can think about an walky talky. With one device who they are connecting to.
I have this code that works with an one to one connection.
import AVFoundation
import UIKit
import WebRTC
import SocketIO
import CoreTelephony
import ReachabilitySwift
let TAG = "ViewController"
let AUDIO_TRACK_ID = TAG + "AUDIO"
let LOCAL_MEDIA_STREAM_ID = TAG + "STREAM"
class ViewController: UIViewController, RTCPeerConnectionDelegate, RTCDataChannelDelegate {
var mediaStream: RTCMediaStream!
var localAudioTrack: RTCAudioTrack!
var remoteAudioTrack: RTCAudioTrack!
var dataChannel: RTCDataChannel!
var dataChannelRemote: RTCDataChannel!
var roomName: String!
override func viewDidLoad() {
super.viewDidLoad()
// Do any additional setup after loading the view, typically from a nib.
initWebRTC();
sigConnect(wsUrl: "http://192.168.1.69:3000");
localAudioTrack = peerConnectionFactory.audioTrack(withTrackId: AUDIO_TRACK_ID)
mediaStream = peerConnectionFactory.mediaStream(withStreamId: LOCAL_MEDIA_STREAM_ID)
mediaStream.addAudioTrack(localAudioTrack)
}
func getRoomName() -> String {
return (roomName == nil || roomName.isEmpty) ? "_defaultroom": roomName;
}
// webrtc
var peerConnectionFactory: RTCPeerConnectionFactory! = nil
var peerConnection: RTCPeerConnection! = nil
var mediaConstraints: RTCMediaConstraints! = nil
var socket: SocketIOClient! = nil
var wsServerUrl: String! = nil
var peerStarted: Bool = false
func initWebRTC() {
RTCInitializeSSL()
peerConnectionFactory = RTCPeerConnectionFactory()
let mandatoryConstraints = ["OfferToReceiveAudio": "true", "OfferToReceiveVideo": "false"]
let optionalConstraints = [ "DtlsSrtpKeyAgreement": "true", "RtpDataChannels" : "true", "internalSctpDataChannels" : "true"]
mediaConstraints = RTCMediaConstraints.init(mandatoryConstraints: mandatoryConstraints, optionalConstraints: optionalConstraints)
}
func connect() {
if (!peerStarted) {
sendOffer()
peerStarted = true
}
}
func hangUp() {
sendDisconnect()
stop()
}
func stop() {
if (peerConnection != nil) {
peerConnection.close()
peerConnection = nil
peerStarted = false
}
}
func prepareNewConnection() -> RTCPeerConnection {
var icsServers: [RTCIceServer] = []
icsServers.append(RTCIceServer(urlStrings: ["stun:stun.l.google.com:19302"], username:"",credential: ""))
let rtcConfig: RTCConfiguration = RTCConfiguration()
rtcConfig.tcpCandidatePolicy = RTCTcpCandidatePolicy.disabled
rtcConfig.bundlePolicy = RTCBundlePolicy.maxBundle
rtcConfig.rtcpMuxPolicy = RTCRtcpMuxPolicy.require
rtcConfig.iceServers = icsServers;
peerConnection = peerConnectionFactory.peerConnection(with: rtcConfig, constraints: mediaConstraints, delegate: self)
peerConnection.add(mediaStream);
let tt = RTCDataChannelConfiguration();
tt.isOrdered = false;
self.dataChannel = peerConnection.dataChannel(forLabel: "testt", configuration: tt)
self.dataChannel.delegate = self
print("Make datachannel")
return peerConnection;
}
// RTCPeerConnectionDelegate - begin [ ///////////////////////////////////////////////////////////////////////////////
/** Called when the SignalingState changed. */
public func peerConnection(_ peerConnection: RTCPeerConnection, didChange stateChanged: RTCSignalingState){
print("signal state: \(stateChanged.rawValue)")
}
/** Called when media is received on a new stream from remote peer. */
public func peerConnection(_ peerConnection: RTCPeerConnection, didAdd stream: RTCMediaStream){
if (peerConnection == nil) {
return
}
if (stream.audioTracks.count > 1) {
print("Weird-looking stream: " + stream.description)
return
}
}
/** Called when a remote peer closes a stream. */
public func peerConnection(_ peerConnection: RTCPeerConnection, didRemove stream: RTCMediaStream){}
/** Called when negotiation is needed, for example ICE has restarted. */
public func peerConnectionShouldNegotiate(_ peerConnection: RTCPeerConnection){}
/** Called any time the IceConnectionState changes. */
public func peerConnection(_ peerConnection: RTCPeerConnection, didChange newState: RTCIceConnectionState){}
/** Called any time the IceGatheringState changes. */
public func peerConnection(_ peerConnection: RTCPeerConnection, didChange newState: RTCIceGatheringState){}
/** New ice candidate has been found. */
public func peerConnection(_ peerConnection: RTCPeerConnection, didGenerate candidate: RTCIceCandidate){
print("iceCandidate: " + candidate.description)
let json:[String: AnyObject] = [
"type" : "candidate" as AnyObject,
"sdpMLineIndex" : candidate.sdpMLineIndex as AnyObject,
"sdpMid" : candidate.sdpMid as AnyObject,
"candidate" : candidate.sdp as AnyObject
]
sigSendIce(msg: json as NSDictionary)
}
/** Called when a group of local Ice candidates have been removed. */
public func peerConnection(_ peerConnection: RTCPeerConnection, didRemove candidates: [RTCIceCandidate]){}
/** New data channel has been opened. */
public func peerConnection(_ peerConnection: RTCPeerConnection, didOpen dataChannel: RTCDataChannel){
print("Datachannel is open, name: \(dataChannel.label)")
dataChannel.delegate = self
self.dataChannelRemote = dataChannel
}
// RTCPeerConnectionDelegate - end ]/////////////////////////////////////////////////////////////////////////////////
public func dataChannel(_ dataChannel: RTCDataChannel, didReceiveMessageWith buffer: RTCDataBuffer){
print("iets ontvangen");
}
public func dataChannelDidChangeState(_ dataChannel: RTCDataChannel){
print("channel.state \(dataChannel.readyState.rawValue)");
}
func sendData(message: String) {
let newData = message.data(using: String.Encoding.utf8)
let dataBuff = RTCDataBuffer(data: newData!, isBinary: false)
self.dataChannel.sendData(dataBuff)
}
func onOffer(sdp:RTCSessionDescription) {
print("on offer shizzle")
setOffer(sdp: sdp)
sendAnswer()
peerStarted = true;
}
func onAnswer(sdp:RTCSessionDescription) {
setAnswer(sdp: sdp)
}
func onCandidate(candidate:RTCIceCandidate) {
peerConnection.add(candidate)
}
func sendSDP(sdp:RTCSessionDescription) {
print("Converting sdp...")
let json:[String: AnyObject] = [
"type" : sdp.type.rawValue as AnyObject,
"sdp" : sdp.sdp.description as AnyObject
]
sigSend(msg: json as NSDictionary);
}
func sendOffer() {
peerConnection = prepareNewConnection();
peerConnection.offer(for: mediaConstraints) { (RTCSessionDescription, Error) in
if(Error == nil){
print("send offer")
self.peerConnection.setLocalDescription(RTCSessionDescription!, completionHandler: { (Error) in
print("Sending: SDP")
print(RTCSessionDescription as Any)
self.sendSDP(sdp: RTCSessionDescription!)
})
} else {
print("sdp creation error: \(Error)")
}
}
}
func setOffer(sdp:RTCSessionDescription) {
if (peerConnection != nil) {
print("peer connection already exists")
}
peerConnection = prepareNewConnection();
peerConnection.setRemoteDescription(sdp) { (Error) in
}
}
func sendAnswer() {
print("sending Answer. Creating remote session description...")
if (peerConnection == nil) {
print("peerConnection NOT exist!")
return
}
peerConnection.answer(for: mediaConstraints) { (RTCSessionDescription, Error) in
print("ice shizzle")
if(Error == nil){
self.peerConnection.setLocalDescription(RTCSessionDescription!, completionHandler: { (Error) in
print("Sending: SDP")
print(RTCSessionDescription as Any)
self.sendSDP(sdp: RTCSessionDescription!)
})
} else {
print("sdp creation error: \(Error)")
}
}
}
func setAnswer(sdp:RTCSessionDescription) {
if (peerConnection == nil) {
print("peerConnection NOT exist!")
return
}
peerConnection.setRemoteDescription(sdp) { (Error) in
print("remote description")
}
}
func sendDisconnect() {
let json:[String: AnyObject] = [
"type" : "user disconnected" as AnyObject
]
sigSend(msg: json as NSDictionary);
}
// websocket related operations
func sigConnect(wsUrl:String) {
wsServerUrl = wsUrl;
print("connecting to " + wsServerUrl)
socket = SocketIOClient(socketURL: NSURL(string: wsServerUrl)! as URL)
socket.on("connect") { data in
print("WebSocket connection opened to: " + self.wsServerUrl);
self.sigEnter();
}
socket.on("disconnect") { data in
print("WebSocket connection closed.")
}
socket.on("message") { (data, emitter) in
if (data.count == 0) {
return
}
let json = data[0] as! NSDictionary
print("WSS->C: " + json.description);
let type = json["type"] as! Int
if (type == RTCSdpType.offer.rawValue) {
print("Received offer, set offer, sending answer....");
let sdp = RTCSessionDescription(type: RTCSdpType(rawValue: type)!, sdp: json["sdp"] as! String)
self.onOffer(sdp: sdp);
} else if (type == RTCSdpType.answer.rawValue && self.peerStarted) {
print("Received answer, setting answer SDP");
let sdp = RTCSessionDescription(type: RTCSdpType(rawValue: type)!, sdp: json["sdp"] as! String)
self.onAnswer(sdp: sdp);
} else {
print("Unexpected websocket message");
}
}
socket.on("ice") { (data, emitter) in
if (data.count == 0) {
return
}
let json = data[0] as! NSDictionary
print("WSS->C: " + json.description);
let type = json["type"] as! String
if (type == "candidate" && self.peerStarted) {
print("Received ICE candidate...");
let candidate = RTCIceCandidate(
sdp: json["candidate"] as! String,
sdpMLineIndex: Int32(json["sdpMLineIndex"] as! Int),
sdpMid: json["sdpMid"] as? String)
self.onCandidate(candidate: candidate);
} else {
print("Unexpected websocket message");
}
}
socket.connect();
}
func sigRecoonect() {
socket.disconnect();
socket.connect();
}
func sigEnter() {
let roomName = getRoomName();
print("Entering room: " + roomName);
socket.emit("enter", roomName);
}
func sigSend(msg:NSDictionary) {
socket.emit("message", msg)
}
func sigSendIce(msg:NSDictionary) {
socket.emit("ice", msg)
}
}
So I thought that I need an array with the peers. And the mediaStream, localAudioTrack and the dataChannel needs to be one object because the local audio is the same? Are there good solutions for this? Because I don't know how to properly implement this.
I am investigating different questions and project referencing to an multi call webrtc setup.
I saw this (website) webrtc setup at GitHub: https://github.com/anoek/webrtc-group-chat-example/blob/master/client.html
I'm going to try to reverse engineer this to swift:). Any help is really appreciated.
As many as you like. You can cram anywhere from one to a million users into a WebRTC call. You've been asked to create a group video call, and obviously, the technology selected for the project was WebRTC.
So no, WebRTC cannot be used with IP multicast.
Building the Python WebRTC Signaling Server First, create a Flask app instance and set a secret_key for it. Then, create a SocketIO instance with this app and start the server on port 9000 from __main__ with socketio. run function.
I would suggest against a one-to-many architecture where a single device needs to send its media to all others. This breaks awfully fast (like after 2-3 devices it needs to connect to).
The reason for that is that uplinks are usually limited in capacity and even when they aren't, devices aren't really geared to streaming so much data to many other devices.
To do what you want at "scale", use a server component that routes media to the other devices. Look at https://jitsi.org/ and http://www.kurento.org/ for starting points.
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