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author | Fabio Alessandrelli <fabio.alessandrelli@gmail.com> | 2019-05-11 01:46:27 +0200 |
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committer | Fabio Alessandrelli <fabio.alessandrelli@gmail.com> | 2019-05-16 11:21:20 +0200 |
commit | 729b1e9941c0eeb0d51608c313ae2096ce13b2ba (patch) | |
tree | 8acc4290e784d0e0a662d7b5aa13f6a92a3c882d /modules/gdnative/gdnative_library_singleton_editor.h | |
parent | eded8d52e3f11357451214ab4d957ed1f7a31b18 (diff) |
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`
Diffstat (limited to 'modules/gdnative/gdnative_library_singleton_editor.h')
0 files changed, 0 insertions, 0 deletions