François Laignel 231389a990 webrtc: only use close() to close websockets ... In the signaller clients and servers, the following sequence is used to close the websocket (in the [send task]): ```rust ws_sink.send(WsMessage::Close(None)).await?; ws_sink.close().await?; ``` tungstenite's [`WebSocket::close()` doc] states: > Calling this function is the same as calling `write(Message::Close(..))`` So we might think they are redundant and either could be used for this purpose (`send()` calls `write()`, then `flush()`). The result is actually is bit different as `write()` starts by checking the state of the connection and [returns `SendAfterClosing`] if the socket is no longer active, which is the case when a closing request has been received from the peer via a [call to `do_close()`]). Note that `do_close()` also enqueues a `Close` frame. This behaviour is visible from the server's logs: ``` 1. tungstenite::protocol: Received close frame: None 2. tungstenite::protocol: Replying to close with Frame { header: FrameHeader { .., opcode: Control(Close), .. }, payload: [] } 3. gst_plugin_webrtc_signalling::server: Received message Ok(Close(None)) 4. gst_plugin_webrtc_signalling::server: connection closed: None this_id=cb13892f-b4d5-4d59-95e2-b3873a7bd319 5. remove_peer{peer_id="cb13892f-b4d5-4d59-95e2-b3873a7bd319"}: gst_plugin_webrtc_signalling::server: close time.busy=285µs time.idle=55.5µs 6. async_tungstenite: websocket start_send error: WebSocket protocol error: Sending after closing is not allowed ``` 1: The server's websocket receives the peer's `Close(None)`. 2: `do_close()` enqueues a `Close` frame. 3: The incoming `Close(None)` is handled by the server. 4 & 5: perform session closing. 6: `ws_sink.send(WsMessage::Close(None))` attempts to `write()` while the ws is no longer active. The error causes an early return, which means that the enqueued `Close` frame is not flushed. Depending on the peer's shutdown sequence, this can result in the following error, which can bubble up as a `Message` on the application's bus: ``` ERROR: from element /GstPipeline:pipeline0/GstWebRTCSrc:webrtcsrc0: GStreamer encountered a general stream error. Additional debug info: net/webrtc/src/webrtcsrc/imp.rs(625): gstrswebrtc::webrtcsrc:👿:BaseWebRTCSrc::connect_signaller::{{closure}}::{{closure}} (): /GstPipeline:pipeline0/GstWebRTCSrc:webrtcsrc0: Signalling error: Error receiving: WebSocket protocol error: Connection reset without closing handshake ``` On the other hand, [`close()` ensures the ws is active] before attempting to write a `Close` frame. If it's not, it only flushes the stream. Thus, when we want to be able to close the websocket and/or to honor the closing handshake in response to the peer `Close` message, the `ws_sink.close()` variant is preferable. This can be verified in the resulting server's logs: ``` tungstenite::protocol: Received close frame: None tungstenite::protocol: Replying to close with Frame { header: FrameHeader { is_final: true, rsv1: false, rsv2: false, rsv3: false, opcode: Control(Close), mask: None}, payload: [] } gst_plugin_webrtc_signalling::server: Received message Ok(Close(None)) gst_plugin_webrtc_signalling::server: connection closed: None this_id=192ed7ff-3b9d-45c5-be66-872cbe67d190 remove_peer{peer_id="192ed7ff-3b9d-45c5-be66-872cbe67d190"}: gst_plugin_webrtc_signalling::server: close time.busy=22.7µs time.idle=37.4µs tungstenite::protocol: Sending pong/close ``` We now get the notification `Sending pong/close` (the closing handshake) instead of `websocket start_send error` from step 6 with previous variant. The `Connection reset without closing handshake` was not observed after this change. [send task]: 63b568f4a0/net/webrtc/signalling/src/server/mod.rs (L165) [`WebSocket::close()` doc]: https://docs.rs/tungstenite/0.21.0/tungstenite/protocol/struct.WebSocket.html#method.close [returns `SendAfterClosing`]: 85463b264e/src/protocol/mod.rs (L437) [call to `do_close()`]: 85463b264e/src/protocol/mod.rs (L601) [`close()` ensures the ws is active]: 85463b264e/src/protocol/mod.rs (L531) Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1456>		2024-02-12 17:50:24 +01:00
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src	webrtc: only use close() to close websockets	2024-02-12 17:50:24 +01:00
Cargo.toml	Update versions to 0.9.12	2023-11-10 17:47:41 +02:00