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`

modules/gdnative/config.py

@@ -16,7 +16,8 @@ def get_doc_classes():
         "ResourceFormatLoaderVideoStreamGDNative",
         "StreamPeerGDNative",
         "VideoStreamGDNative",
-        "WebRTCPeerGDNative",
+        "WebRTCPeerConnectionGDNative",
+        "WebRTCDataChannelGDNative",
     ]
 
 def get_doc_path():