virtualx-engine/modules/gdnative/doc_classes/WebRTCPeerConnectionGDNative.xml
Fabio Alessandrelli 729b1e9941 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`
2019-05-16 11:21:20 +02:00

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XML

<?xml version="1.0" encoding="UTF-8" ?>
<class name="WebRTCPeerConnectionGDNative" inherits="WebRTCPeerConnection" category="Core" version="3.2">
<brief_description>
</brief_description>
<description>
</description>
<tutorials>
</tutorials>
<methods>
</methods>
<constants>
</constants>
</class>