Commit graph

37 commits

Author SHA1 Message Date
Sebastian Dröge
b2d5ee48cd Update to async-tungstenite 0.25
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1455>
2024-02-11 11:31:24 +02:00
Bilal Elmoussaoui
0615a16124 Use workspace features for crates metadata/deps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1446>
2024-02-05 15:34:31 +01:00
François Laignel
91bfd0f7c3 webrtc: signallers: attempt to close the ws when an error occurs
This commit discards the early error returns in the send tasks to log the error
and attempt to close the websocket.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1435>
2024-02-01 18:08:41 +01:00
François Laignel
f54d714afd 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state of the connection and [returns `SendAfterClosing`] if the socket is no
longer active, which is the case when a closing request has been received from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/1435>
2024-02-01 18:08:41 +01:00
Nirbheek Chauhan
2d85048925 webrtc/signalling: We get the address when accepting
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1412>
2023-12-29 13:28:48 +00:00
Nirbheek Chauhan
63b568f4a0 webrtc/signalling: Fix potential hang and FD leak
If a peer connects via TCP and never initiates TLS, then the server
will get stuck in the accept loop. Spawn a task when accepting a TLS
connection, and timeout if it doesn't complete in 5 seconds.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1412>
2023-12-29 13:28:48 +00:00
Sebastian Dröge
181bd13103 Update to async-tungstenite 0.24
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
2023-12-09 12:17:11 +02:00
Sebastian Dröge
557b249e11 Update to AWS SDK 0.34 and tracing-log 0.2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1374>
2023-10-27 10:19:15 +03:00
Sebastian Dröge
4569b7eca6 Fix various new 1.73 clippy warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1347>
2023-10-03 17:47:30 +03:00
Sebastian Dröge
d688aeb184 Update versions to 0.12.0-alpha.1 2023-08-10 17:21:11 +03:00
Sebastian Dröge
b3826c108d webrtc: Update to async-tungstenite 0.23
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1299>
2023-08-09 13:18:44 +03:00
Sebastian Dröge
31b1cb8ca6 Update minimum supported Rust version to 1.70
gtk-rs will update soonish too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1280>
2023-07-19 09:19:34 +03:00
Bilal Elmoussaoui
dd2d7d9215 Use re-exported once_cell
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1268>
2023-07-06 17:50:49 +03:00
Mathieu Duponchelle
7447d95f1b webrtc/signalling: fix race condition in message ordering
Spawning one task per message to send out instead of sending them out
sequentially from the one task used to poll the handler sometimes
resulted in peers receiving ICE candidates before SDP offers, triggering
hard to understand errors in the browser.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1236>
2023-06-08 13:24:45 +02:00
Sebastian Dröge
cb5b527d74 Update to AWS SDK 0.27 and async-tungstenite 0.22
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1199>
2023-05-02 15:30:00 +03:00
Sebastian Dröge
5451035215 Update async-tungstenite and AWS SDK dependencies
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1187>
2023-04-21 10:48:10 +00:00
Thibault Saunier
ce3bb2f1d4 Add a webrtcsrc element
Updating the docker image to include:
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3236

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/932>
2023-02-28 20:50:15 -03:00
Thibault Saunier
4ec441560b webrtc: Enhance debug messages when using unknown peer ID
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/932>
2023-02-28 19:28:51 +00:00
Sebastian Dröge
9fc1404415 Update minimum supported Rust version to 1.66
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1096>
2023-02-20 11:09:01 +02:00
Sebastian Dröge
ac8afc4ac0 Update to async-tungstenite 0.20
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1087>
2023-02-10 13:03:07 +02:00
Sebastian Dröge
1e13dbb99c Update versions to 0.11.0-alpha.1 2023-02-10 00:23:56 +02:00
Sebastian Dröge
3b4c48d9f5 Fix various new clippy warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1062>
2023-01-25 10:31:19 +02:00
Zhao, Gang
9fa838e366 webrtc: Fix rustfmt errors
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1019>
2022-12-27 11:12:54 +02:00
Zhao, Gang
1ffeb4d44d webrtc: Move from async-std to tokio
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1019>
2022-12-26 23:10:40 +00:00
Sebastian Dröge
fb42cd8a0f net: Update to async-tungstenite 0.19
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1005>
2022-12-11 12:54:24 +02:00
Sebastian Dröge
bf6bdab80c webrtc: Remove version requirement from internal crate dependencies 2022-10-24 19:50:24 +03:00
Sebastian Dröge
f2223cf2cb Update versions to 0.10.0-alpha.1 2022-10-24 19:31:19 +03:00
Sebastian Dröge
b64f951160 Update to async-tungstenite 0.18 2022-10-24 18:03:33 +03:00
Thibault Saunier
5c89c3db69 webrtc: Rename and add to meson build the signalling server
The binary was only called `server` it has been renamed to
`gst-webrtc-signalling-server` and is installed in meson.
2022-10-20 18:20:15 +00:00
Sebastian Dröge
c0bf05d4bb webrtc: Minor cleanup 2022-10-20 13:20:32 +03:00
Thibault Saunier
71ed04d89b webrtc: Rename signaller and protocol crates 2022-10-20 13:32:31 +02:00
Thibault Saunier
0f0dec7fa9 webrtc: Fix fmt issues 2022-10-20 11:51:59 +02:00
Thibault Saunier
5ab7be6124 webrtc: Add SDPX license header on every file 2022-10-20 11:51:58 +02:00
Thibault Saunier
39c0dcb0d4 Plug webrtc in 2022-10-20 11:51:58 +02:00
Thibault Saunier
b164daf510 webrtc: Fix clippy issues 2022-10-20 11:51:58 +02:00
Thibault Saunier
87fd49a9bf webrtc:signalling: Remove short option for 'host' in the cli
It clashes with `--help`
2022-10-20 11:51:58 +02:00
Thibault Saunier
5e7537953c webrtc: Move to net/webrtc 2022-10-18 15:18:53 +02:00