WebRTC refactor. Data channels, STUN/TURN support.

A big refactor to the WebRTC module. API is now considered quite stable.

Highlights:

- Renamed `WebRTCPeer` to `WebRTCPeerConnection`.
- `WebRTCPeerConnection` no longer act as `PacketPeer`, it only handle the connection itself (a bit like `TCP_Server`)
- Added new `WebRTCDataChannel` class which inherits from `PacketPeer` to handle data transfer.
- Add `WebRTCPeerConnection.initialize` method to create a new connection with the desired configuration provided as dictionary ([see MDN docs](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/RTCPeerConnection#RTCConfiguration_dictionary)).
- Add `WebRTCPeerConnection.create_data_channel` method to create a data channel for the given connection. The connection must be in `STATE_NEW` as specified by the standard ([see MDN docs for options](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/createDataChannel#RTCDataChannelInit_dictionary)).
- Add a `data_channel_received` signal to `WebRTCPeerConnection` for in-band (not negotiated) channels.
- Renamed `WebRTCPeerConnection` `offer_created` signal to `session_description_created`.
- Renamed `WebRTCPeerConnection` `new_ice_candidate` signal to `ice_candidate_created`

1 files changed, 66 insertions, 0 deletions

diff --git a/modules/webrtc/webrtc_peer_connection_js.h b/modules/webrtc/webrtc_peer_connection_js.h
new file mode 100644
index 0000000000..43c0e3d6ee
--- /dev/null
+++ b/modules/webrtc/webrtc_peer_connection_js.h
@@ -0,0 +1,66 @@
+/*************************************************************************/
+/*  webrtc_peer_connection_js.h                                          */
+/*************************************************************************/
+/*                       This file is part of:                           */
+/*                           GODOT ENGINE                                */
+/*                      https://godotengine.org                          */
+/*************************************************************************/
+/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur.                 */
+/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md)    */
+/*                                                                       */
+/* Permission is hereby granted, free of charge, to any person obtaining */
+/* a copy of this software and associated documentation files (the       */
+/* "Software"), to deal in the Software without restriction, including   */
+/* without limitation the rights to use, copy, modify, merge, publish,   */
+/* distribute, sublicense, and/or sell copies of the Software, and to    */
+/* permit persons to whom the Software is furnished to do so, subject to */
+/* the following conditions:                                             */
+/*                                                                       */
+/* The above copyright notice and this permission notice shall be        */
+/* included in all copies or substantial portions of the Software.       */
+/*                                                                       */
+/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,       */
+/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF    */
+/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
+/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY  */
+/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,  */
+/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE     */
+/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.                */
+/*************************************************************************/
+
+#ifndef WEBRTC_PEER_CONNECTION_JS_H
+#define WEBRTC_PEER_CONNECTION_JS_H
+
+#ifdef JAVASCRIPT_ENABLED
+
+#include "webrtc_peer_connection.h"
+
+class WebRTCPeerConnectionJS : public WebRTCPeerConnection {
+
+private:
+	int _js_id;
+	ConnectionState _conn_state;
+
+public:
+	static WebRTCPeerConnection *_create() { return memnew(WebRTCPeerConnectionJS); }
+	static void make_default() { WebRTCPeerConnection::_create = WebRTCPeerConnectionJS::_create; }
+
+	void _on_connection_state_changed();
+	virtual ConnectionState get_connection_state() const;
+
+	virtual Error initialize(Dictionary configuration = Dictionary());
+	virtual Ref<WebRTCDataChannel> create_data_channel(String p_channel_name, Dictionary p_channel_config = Dictionary());
+	virtual Error create_offer();
+	virtual Error set_remote_description(String type, String sdp);
+	virtual Error set_local_description(String type, String sdp);
+	virtual Error add_ice_candidate(String sdpMidName, int sdpMlineIndexName, String sdpName);
+	virtual Error poll();
+	virtual void close();
+
+	WebRTCPeerConnectionJS();
+	~WebRTCPeerConnectionJS();
+};
+
+#endif
+
+#endif // WEBRTC_PEER_CONNECTION_JS_H