public SymplePlayerEngineWebRTC(SymplePlayer player) : base(player)
{
Messenger.Broadcast(SympleLog.LogInfo, "symple:webrtc: init");
#if NETFX_CORE
if (!webrtcInitialized)
{
WebRTC.Initialize(null); // needed before calling any webrtc functions http://stackoverflow.com/questions/43331677/webrtc-for-uwp-new-rtcpeerconnection-doesnt-complete-execution
webrtcInitialized = true;
}
if (player.options.rtcConfig != null)
{
this.rtcConfig = player.options.rtcConfig;
}
else
{
this.rtcConfig = new RTCConfiguration();
this.rtcConfig.IceServers.Add(new RTCIceServer()
{
Url = "stun:stun.l.google.com:19302", Username = string.Empty, Credential = string.Empty
});
}
#endif
/*
* this.rtcOptions = player.options.rtcOptions || {
* optional: [
* { DtlsSrtpKeyAgreement: true } // required for FF <=> Chrome interop
* ]
* };
*/
// Specifies that this client will be the ICE initiator,
// and will be sending the initial SDP Offer.
this.initiator = player.options.initiator;
Messenger.Broadcast(SympleLog.LogInfo, "symple:webrtc: constructor, set this.initiator to " + this.initiator);
#if NETFX_CORE
// Reference to the active local or remote media stream
this.activeStream = null;
#endif
}
public void initAndStartWebRTC()
{
#if NETFX_CORE
JObject CLIENT_OPTIONS = new JObject();
CLIENT_OPTIONS["secure"] = true;
CLIENT_OPTIONS["url"] = this.SignallingServerUrl;
CLIENT_OPTIONS["peer"] = new JObject();
CLIENT_OPTIONS["peer"]["user"] = this.LocalPeerUsername;
CLIENT_OPTIONS["peer"]["name"] = this.LocalPeerNameLabel;
CLIENT_OPTIONS["peer"]["group"] = this.LocalPeerGroup;
SymplePlayerOptions playerOptions = new SymplePlayerOptions();
playerOptions.engine = "WebRTC";
switch (UserType)
{
case StarUserType.TRAINEE:
playerOptions.initiator = true;
break;
case StarUserType.MENTOR:
playerOptions.initiator = false;
break;
default:
break;
}
// WebRTC config
// This is where you would add TURN servers for use in production
RTCConfiguration WEBRTC_CONFIG = new RTCConfiguration
{
IceServers = new List <RTCIceServer> {
new RTCIceServer {
Url = "stun:stun.l.google.com:19302", Username = string.Empty, Credential = string.Empty
},
new RTCIceServer {
Url = "stun:stun1.l.google.com:19302", Username = string.Empty, Credential = string.Empty
},
new RTCIceServer {
Url = "stun:stun2.l.google.com:19302", Username = string.Empty, Credential = string.Empty
},
new RTCIceServer {
Url = "stun:stun3.l.google.com:19302", Username = string.Empty, Credential = string.Empty
},
new RTCIceServer {
Url = "stun:stun4.l.google.com:19302", Username = string.Empty, Credential = string.Empty
},
new RTCIceServer {
Url = "turn:numb.viagenie.ca", Username = "******", Credential = "0O@S&YfP$@56"
}
}
};
playerOptions.rtcConfig = WEBRTC_CONFIG;
//playerOptions.iceMediaConstraints = asdf; // TODO: not using iceMediaConstraints in latest code?
playerOptions.onStateChange = (player, state, message) =>
{
player.displayStatus(state);
};
Messenger.Broadcast(SympleLog.LogInfo, "creating player");
player = new SymplePlayer(playerOptions);
Messenger.Broadcast(SympleLog.LogInfo, "creating client");
client = new SympleClient(CLIENT_OPTIONS);
client.on("announce", (peer) => {
Messenger.Broadcast(SympleLog.LogInfo, "Authentication success: " + peer);
});
client.on("addPeer", (peerObj) =>
{
JObject peer = (JObject)peerObj;
Messenger.Broadcast(SympleLog.LogInfo, "adding peer: " + peer);
if (this.UserType == StarUserType.TRAINEE)
{
// the TRAINEE user waits for a peer with a specific username, then once it's connected it automatically starts sending video
if ((string)peer["user"] == this.ExpectedRemoteReceiverUsername && !initialized)
{
initialized = true;
remotePeer = peer;
startPlaybackAndRecording();
}
}
});
client.on("presence", (presence) =>
{
Messenger.Broadcast(SympleLog.LogInfo, "Recv presence: " + presence);
});
client.on("removePeer", (peerObj) =>
{
JObject peer = (JObject)peerObj;
Messenger.Broadcast(SympleLog.LogInfo, "Removing peer: " + peer);
if (remotePeer != null && remotePeer["id"].Equals(peer["id"]))
{
initialized = false;
remotePeer = null;
player.engine.destroy();
player.engine = null;
}
});
client.on("message", (mObj) =>
{
Messenger.Broadcast(SympleLog.LogTrace, "mObj.GetType().ToString(): " + mObj.GetType().ToString());
JObject m = (JObject)((Object[])mObj)[0];
Messenger.Broadcast(SympleLog.LogTrace, "recv message: " + m);
Messenger.Broadcast(SympleLog.LogTrace, "remotePeer: " + remotePeer);
var mFrom = m["from"];
JToken mFromId = null;
if (mFrom.Type == JTokenType.Object)
{
mFromId = mFrom["id"];
}
if (remotePeer != null && !remotePeer["id"].Equals(mFromId))
{
Messenger.Broadcast(SympleLog.LogInfo, "Dropping message from unknown peer: " + m);
return;
}
if (m["offer"] != null)
{
switch (UserType)
{
case StarUserType.TRAINEE:
Messenger.Broadcast(SympleLog.LogInfo, "Unexpected offer for one-way streaming");
break;
case StarUserType.MENTOR:
Messenger.Broadcast(SympleLog.LogInfo, "Receive offer: " + m["offer"]);
remotePeer = (JObject)m["from"];
JObject playParams = new JObject(); // empty params
player.play(playParams);
var engine = (SymplePlayerEngineWebRTC)player.engine;
engine.recvRemoteSDP((JObject)m["offer"]);
engine.sendLocalSDP = (desc) =>
{
Messenger.Broadcast(SympleLog.LogInfo, "Send answer: " + desc);
JObject sessionDesc = new JObject();
sessionDesc["sdp"] = desc.Sdp;
if (desc.Type == Org.WebRtc.RTCSdpType.Answer)
{
sessionDesc["type"] = "answer";
}
else if (desc.Type == Org.WebRtc.RTCSdpType.Offer)
{
sessionDesc["type"] = "offer";
}
else if (desc.Type == Org.WebRtc.RTCSdpType.Pranswer)
{
sessionDesc["type"] = "pranswer";
}
JObject parameters = new JObject();
parameters["to"] = remotePeer;
parameters["type"] = "message";
parameters["answer"] = sessionDesc;
client.send(parameters);
};
engine.sendLocalCandidate = (cand) =>
{
JObject candidateObj = new JObject();
candidateObj["candidate"] = cand.Candidate;
candidateObj["sdpMid"] = cand.SdpMid;
candidateObj["sdpMLineIndex"] = cand.SdpMLineIndex;
JObject parameters = new JObject();
parameters["to"] = remotePeer;
parameters["type"] = "message";
parameters["candidate"] = candidateObj;
client.send(parameters);
};
break;
default:
break;
}
}
else if (m["answer"] != null)
{
switch (UserType)
{
case StarUserType.TRAINEE:
SymplePlayerEngineWebRTC engine = (SymplePlayerEngineWebRTC)player.engine;
string answerJsonString = JsonConvert.SerializeObject(m["answer"], Formatting.None);
JObject answerParams = (JObject)m["answer"];
Messenger.Broadcast(SympleLog.LogTrace, "Receive answer: " + answerJsonString);
engine.recvRemoteSDP(answerParams);
break;
case StarUserType.MENTOR:
Messenger.Broadcast(SympleLog.LogInfo, "Unexpected answer for one-way streaming");
break;
default:
break;
}
}
else if (m["candidate"] != null)
{
SymplePlayerEngineWebRTC engine = (SymplePlayerEngineWebRTC)player.engine;
JObject candidateParams = (JObject)m["candidate"];
Messenger.Broadcast(SympleLog.LogInfo, "Using Candidate: " + candidateParams);
engine.recvRemoteCandidate(candidateParams);
}
});
client.on("disconnect", (peer) =>
{
Messenger.Broadcast(SympleLog.LogInfo, "Disconnected from server");
});
client.on("error", (error) =>
{
Messenger.Broadcast(SympleLog.LogError, "Connection error: " + error);
});
client.connect();
#else
Messenger.Broadcast(SympleLog.LogInfo, "not actually connecting via webrtc because NETFX_CORE not defined (probably this is in the unity editor)");
#endif
}
public SymplePlayerEngine(SymplePlayer player)
{
this.player = player;
this.fps = 0;
this.seq = 0;
}