Commit graph

210 commits

Author SHA1 Message Date
Maksym Khomenko
2228f882d8 webrtcsink: apply rustfmt
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1469>
2024-02-20 19:29:28 +02:00
Xavier Claessens
2572afbf15 janusvr: Add secret-key property
Every API calls have an optional "apisecret" argument.

Sponsored-by: Netflix Inc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1465>
2024-02-16 14:04:59 +00:00
Jordan Yelloz
67b7cf9764 webrtcsink: Added sinkpad with "msid" property
This forwards to the webrtcbin sinkpad's msid when specified.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1442>
2024-02-12 15:04:44 +00:00
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
Sebastian Dröge
91abc62ad0 webrtcsink: Fix new clippy warning
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1445>
2024-02-05 12:53:20 +02:00
Sebastian Dröge
1a55c70114 Switch git dependencies to explicitly name branch
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1445>
2024-02-05 12:51:36 +02:00
Sebastian Dröge
ffa830ae9b Update for GLib prelude re-organization
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1444>
2024-02-03 12:30:15 +02:00
Jordan Yelloz
311fda649f livekit_signaller: Added high-quality layer for video streams
This change addresses a cosmetic issue with livekit, where the
connection quality indicator seen by other users shows bad quality
unless the track is added with a high quality layer. The details of the
layer submitted aren't significant for this purpose.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1443>
2024-02-02 20:57:17 +00:00
Robert Ayrapetyan
916a8b959e doc: add http headers for webrtcsink signaller
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1419>
2024-02-01 19:31:58 +00:00
Robert Ayrapetyan
972b9e5474 doc: add docstrings for signaller object
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1419>
2024-02-01 19:31:58 +00:00
Robert Ayrapetyan
7a72b2fc25 webrtcsink-signalling: add headers support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1419>
2024-02-01 19:31:58 +00:00
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
Taruntej Kanakamalla
50e905fe4b webrtc: conditional compile for features with 1_22 dependency
Few features being used in webrtcsink like
the signal `request-aux-sender` are introduced
to webrtcbin in gstreamer release 1.22.

Rename the feature gst1_22 to v1_22 for uniformity.

Add v1_22 to default features.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1241>
2024-02-01 15:08:11 +05:30
Sebastian Dröge
4ad101b53b Use once_cell crate directly again
The glib crate does not depend on it anymore and also does not re-export
it anymore.

Also switch some usages of OnceCell to OnceLock from std.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1441>
2024-01-31 18:07:57 +02:00
Sebastian Dröge
451d928026 webrtc: Update AWS signaller to http 1
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1441>
2024-01-31 18:07:57 +02:00
Sebastian Dröge
764143d971 webrtc: Remove unnecessary manual Send+Sync implementations for signallers
These are automatically implemented.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/483

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1432>
2024-01-18 10:01:25 +02:00
Sebastian Dröge
1af18f3028 webrtc: Require Send+Sync for signaller implementations
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1432>
2024-01-18 10:01:01 +02:00
Eva Pace
80b58f3b45 net/webrtc/janusvr: add JanusVRWebRTCSink plugin/signaller
The JanusVRWebRTCSink is a new plugin that integrates with the Video
Room plugin of the Janus Gateway, which simplifies WebRTC communication.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1362>
2024-01-17 20:33:57 +00:00
Maksym Khomenko
773ebc7854 webrtcsrc: don't restrict RTP extensions to TWCC only
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1381>
2024-01-17 07:34:01 +00:00
Sebastian Dröge
dfa95d8ed3 webrtc: Update to livekit-api / livekit-protocol 0.3
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1427>
2024-01-16 07:52:48 +00:00
Maksym Khomenko
fecbe01e06 webrtcsink: make 'extensions' property usage conditional
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1423>
2024-01-16 07:13:56 +00:00
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
Maksym Khomenko
17f0b61576 webrtcsink: add payloader-setup signal
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1389>
2023-12-23 08:02:08 +00:00
Sebastian Dröge
81dd45c814 webrtc: Downgrade aws-smithy-http to 0.60
Version 0.61 was yanked from crates.io.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1407>
2023-12-14 09:11:07 +02:00
Sebastian Dröge
2f2bf6ca8f webrtc: Update to aws-smithy-http 0.61
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
2023-12-09 12:21:38 +02:00
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
Guillaume Desmottes
6dfd1c1496 use new debug and parse API
Changes from https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1355

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1403>
2023-12-04 15:58:21 +01:00
Sebastian Dröge
f13574d8ed Update further AWS SDK crates to 1.0
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1400>
2023-11-26 10:26:02 +02:00
Mathieu Duponchelle
cf1c7600a2 webrtcsink: don't panic on failure to request pad from webrtcbin
webrtcbin will refuse pad requests for all sorts of reasons, and should
be logging an error when doing so, simply post an error message and let
the application deal with it, the reason for the refusal should
hopefully be available in the logs to the user.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1399>
2023-11-24 19:53:38 +01:00
Sebastian Dröge
c3ced8c7e6 Update to AWS SDK 1.0 / 0.60 / 0.39
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1397>
2023-11-21 10:32:59 +02:00
Sebastian Dröge
1d9c89e3fe Update to AWS SDK 0.101 / 0.59 / 0.38 2023-11-20 10:13:13 +02:00
Taruntej Kanakamalla
43ee6bfc1c net/webrtc: add whipserversrc
Implement new signaller WhipServerSignaller
 - an http server using 'warp'
 - handlers for the POST, OPTIONS, PATCH and DELETE
 - fixed path `/whip/endpoint` as the URI
 - fixed value 'whip-client' as the producer peer id
 - fixed resource url `/whip/resource/whip-client`

Derive whipserversrc element from BaseWebRTCSrc
 - implement constructed method for ObjectImpl to set
  non-default signaller, i.e., WhipServerSignaller
 - bind the properties stun-server and turn-servers to those on
   the Signaller

Connect to 'webrtcbin-ready' signal in the constructor of WhipServerSignaller
 - it will be emitted by the webrtcsrc when the webrtcbin element is ready
 - the closure for this signal will in turn connect to webrtcbin's ice-gathering-state
   and perform send with the answer sdp via the channel
 - the WhipServer will hold its HTTP response in POST handler until this signal
   is received or timeout which happens early

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-11-17 18:08:44 +00:00
Taruntej Kanakamalla
ed3aa740be net/webrtc: deprecate consumer-added on the signaller
add a new signal webrtcbin-ready in this place doing same
thing but can be used for both consumers and producers

Please note this change is only to the consumer-added
signal on the signaller interface.
The consumer-added signal on the webrtcsink is unchanged

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-11-17 18:08:44 +00:00
Taruntej Kanakamalla
2d3d03b4d3 net/webrtc: rename WhipSignaller as WhipClientSignaller
remove generalized names to accommodate for the WhipServer
- name the Signaller for whipsink as WhipClient
- name the Settings for whipsink as WhipClientSettings
- name the State for whipsink as WhipClientState

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-11-17 18:08:44 +00:00
Taruntej Kanakamalla
a0638ec983 net/webrtc: Extract BaseWebRTCSrc
Define a Base for all the webrtcsrc type elements
so they can all be derived from it. Similar to base
element defined for webrtcsink type elements

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-11-17 18:08:44 +00:00
Sebastian Dröge
dee27e35b7 Update to latest AWS SDK
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1395>
2023-11-17 11:22:29 +02:00
Sebastian Dröge
58723f2a8c Update to AWS SDK 0.36
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1394>
2023-11-15 17:20:58 +02:00
Maksym Khomenko
e5fd2c3568 webrtcsrc: add turn-servers property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1380>
2023-11-04 10:19:45 +00:00
Mathieu Duponchelle
5371eb52ad Port to AWS SDK 0.57/0.35
Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1379>
2023-11-03 15:13:45 +00:00
Sebastian Dröge
16b917abb1 Update for gst::Rank API changes 2023-11-02 14:10:59 +02:00
Piotr Brzeziński
436b6d8efb gstwebrtc-api: Patch webrtc-adapter to fix Safari behaviour
There's currently a Safari-side bug causing webrtc-adapter to be unable to correctly shim the empty-candidate scenario
which we're using. This patch is very much a workaround and should be removed as soon as Safari and/or webrtc-adapter
fixes this on their side.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/439
https://github.com/webrtcHacks/adapter/issues/1140

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1377>
2023-10-30 16:36:11 +00:00
Sebastian Dröge
855b03a9ea Use let-else instead of match for weak reference upgrades
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1375>
2023-10-30 11:34:35 +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
2ce04c6a78 webrtc: Update to livekit 0.2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1293>
2023-10-18 10:30:59 +03:00
Sebastian Dröge
d468e1e4a6 Clean up usage of pad probes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1361>
2023-10-17 08:44:06 +03:00
François Laignel
50dd519c4f net/webrtcsrc: define signaller property as CONSTRUCT_ONLY
The "signaller" property used to be defined as MUTABLE_READY which meant that
the property was always set after `constructed()` was called.

Since `connect_signaller()` was called from `constructed()`, only the default
signaller was used.

This commit sets the "signaller" property as CONSTRUCT_ONLY. Using a builder,
this property will now be set before the call to `constructed()`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1324>
2023-10-12 17:38:09 +00:00
François Laignel
785c9557c8 net/webrtcsink: drop State lock before calling set-local-description
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1325>
2023-10-12 15:45:58 +00:00