Merge pull request #845 from nazar-pc/rust-webrtc-server

Add `WebRtcServer` support for Rust

.github/workflows/mediasoup-node.yaml

@@ -6,16 +6,19 @@ jobs:
   ci:
     strategy:
       matrix:
-        # Just 1 OS should be fine for TypeScript.
-        os:
-          - ubuntu-22.04
-          # - macos-11.0
-          # - windows-2019
-        node:
-          - 14
-          - 16
-          - 18
-    runs-on: ${{ matrix.os }}
+        # Different Node versions on Ubuntu, the latest Node on other platforms
+        ci:
+          - os: ubuntu-22.04
+            node: 14
+          - os: ubuntu-22.04
+            node: 16
+          - os: ubuntu-22.04
+            node: 18
+          - os: macos-12
+            node: 18
+          - os: windows-2022
+            node: 18
+    runs-on: ${{ matrix.ci.os }}
 
     steps:
       - name: Checkout
@@ -24,18 +27,19 @@ jobs:
       - name: Node.js
         uses: actions/setup-node@v2
         with:
-          node-version: ${{ matrix.node }}
+          node-version: ${{ matrix.ci.node }}
 
       - name: Configure cache
         uses: actions/cache@v2
         with:
           path: |
             ~/.npm
-          key: ${{ matrix.os }}-node-${{ hashFiles('**/package.json') }}
+          key: ${{ matrix.ci.os }}-node-${{ hashFiles('**/package.json') }}
           restore-keys: |
-            ${{ matrix.os }}-node-
+            ${{ matrix.ci.os }}-node-
 
       - run: npm install
       - run: npm run lint:node
+        if: runner.os == 'Linux'
       - run: npm run typescript:build
       - run: npm run test:node

node/tests/test-WebRtcServer.js

@@ -143,7 +143,7 @@ test('worker.createWebRtcServer() with unavailable listenInfos rejects with Erro
 				},
 				{
 					protocol    : 'udp',
-					ip          : '0.0.0.0',
+					ip          : '127.0.0.1',
 					announcedIp : '1.2.3.4',
 					port        : port1
 				}
@@ -165,7 +165,7 @@ test('worker.createWebRtcServer() with unavailable listenInfos rejects with Erro
 		});
 
 	// Using the same UDP port in a second Worker.
-	await expect(worker1.createWebRtcServer(
+	await expect(worker2.createWebRtcServer(
 		{
 			listenInfos :
 			[

npm-scripts.js

@@ -27,7 +27,7 @@ switch (task)
 		}
 		else
 		{
-			execute('rmdir /s /q node/lib');
+			execute('rmdir /s /q "node/lib"');
 		}
 
 		execute('tsc --project node');
@@ -46,7 +46,7 @@ switch (task)
 		}
 		else
 		{
-			execute('rmdir /s /q node/lib');
+			execute('rmdir /s /q "node/lib"');
 		}
 
 		const watch = new TscWatchClient();

rust/CHANGELOG.md

@@ -8,6 +8,7 @@
   * nlohmann_json 3.10.5.
   * usrsctp snapshot 4e06feb01cadcd127d119486b98a4bd3d64aa1e7.
   * wingetopt 1.00.
+* `WebRtcServer`: A new class that brings to `WebRtcTransports` the ability to listen on a single UDP/TCP port (PR #834, PR #845).
 
 # 0.9.3
 

rust/Cargo.toml

@@ -4,7 +4,7 @@ version = "0.9.3"
 description = "Cutting Edge WebRTC Video Conferencing in Rust"
 categories = ["api-bindings", "multimedia", "network-programming"]
 authors = ["Nazar Mokrynskyi <[email protected]>"]
-edition = "2018"
+edition = "2021"
 license = "ISC"
 keywords = ["webrtc", "ortc", "sfu"]
 documentation = "https://docs.rs/mediasoup"
@@ -70,6 +70,7 @@ actix-web-actors = "4.1.0"
 async-io = "1.6.0"
 criterion = "0.3.5"
 env_logger = "0.9.0"
+portpicker = "0.1.1"
 
 [dev-dependencies.actix-web]
 default-features = false

rust/examples/echo.rs

@@ -179,11 +179,10 @@ impl EchoConnection {
         // We know that for echo example we'll need 2 transports, so we can create both right away.
         // This may not be the case for real-world applications or you may create this at a
         // different time and/or in different order.
-        let transport_options =
-            WebRtcTransportOptions::new(TransportListenIps::new(TransportListenIp {
-                ip: "127.0.0.1".parse().unwrap(),
-                announced_ip: None,
-            }));
+        let transport_options = WebRtcTransportOptions::new(TransportListenIps::new(ListenIp {
+            ip: "127.0.0.1".parse().unwrap(),
+            announced_ip: None,
+        }));
         let producer_transport = router
             .create_webrtc_transport(transport_options.clone())
             .await

rust/examples/multiopus.rs

@@ -144,7 +144,7 @@ impl EchoConnection {
         // For simplicity we will create plain transport for audio producer right away
         let plain_transport = router
             .create_plain_transport({
-                let mut options = PlainTransportOptions::new(TransportListenIp {
+                let mut options = PlainTransportOptions::new(ListenIp {
                     ip: "127.0.0.1".parse().unwrap(),
                     announced_ip: None,
                 });
@@ -225,7 +225,7 @@ impl EchoConnection {
         // this at a different time and/or in different order.
         let consumer_transport = router
             .create_webrtc_transport(WebRtcTransportOptions::new(TransportListenIps::new(
-                TransportListenIp {
+                ListenIp {
                     ip: "127.0.0.1".parse().unwrap(),
                     announced_ip: None,
                 },

rust/examples/svc-simulcast.rs

@@ -199,11 +199,10 @@ impl SvcSimulcastConnection {
         // right away.
         // This may not be the case for real-world applications or you may create this at a
         // different time and/or in different order.
-        let transport_options =
-            WebRtcTransportOptions::new(TransportListenIps::new(TransportListenIp {
-                ip: "127.0.0.1".parse().unwrap(),
-                announced_ip: None,
-            }));
+        let transport_options = WebRtcTransportOptions::new(TransportListenIps::new(ListenIp {
+            ip: "127.0.0.1".parse().unwrap(),
+            announced_ip: None,
+        }));
         let producer_transport = router
             .create_webrtc_transport(transport_options.clone())
             .await

rust/examples/videoroom.rs

@@ -494,7 +494,7 @@ mod participant {
             // right away. This may not be the case for real-world applications or you may create
             // this at a different time and/or in different order.
             let transport_options =
-                WebRtcTransportOptions::new(TransportListenIps::new(TransportListenIp {
+                WebRtcTransportOptions::new(TransportListenIps::new(ListenIp {
                     ip: "127.0.0.1".parse().unwrap(),
                     announced_ip: None,
                 }));

rust/src/data_structures.rs

@@ -50,7 +50,7 @@ impl AppData {
 /// If you use "0.0.0.0" or "::" as ip value, then you need to also provide `announced_ip`.
 #[derive(Debug, Copy, Clone, Ord, PartialOrd, Eq, PartialEq, Hash, Deserialize, Serialize)]
 #[serde(rename_all = "camelCase")]
-pub struct TransportListenIp {
+pub struct ListenIp {
     /// Listening IPv4 or IPv6.
     pub ip: IpAddr,
     /// Announced IPv4 or IPv6 (useful when running mediasoup behind NAT with private IP).
@@ -111,7 +111,7 @@ pub enum IceCandidateTcpType {
 /// Transport protocol.
 #[derive(Debug, Copy, Clone, Ord, PartialOrd, Eq, PartialEq, Hash, Deserialize, Serialize)]
 #[serde(rename_all = "lowercase")]
-pub enum TransportProtocol {
+pub enum Protocol {
     /// TCP.
     Tcp,
     /// UDP.
@@ -130,7 +130,7 @@ pub struct IceCandidate {
     /// The IP address of the candidate.
     pub ip: IpAddr,
     /// The protocol of the candidate.
-    pub protocol: TransportProtocol,
+    pub protocol: Protocol,
     /// The port for the candidate.
     pub port: u16,
     /// The type of candidate (always `Host`).
@@ -182,7 +182,7 @@ pub enum TransportTuple {
         /// Remote port.
         remote_port: u16,
         /// Protocol
-        protocol: TransportProtocol,
+        protocol: Protocol,
     },
     /// Transport tuple without remote endpoint info.
     #[serde(rename_all = "camelCase")]
@@ -192,7 +192,7 @@ pub enum TransportTuple {
         /// Local port.
         local_port: u16,
         /// Protocol
-        protocol: TransportProtocol,
+        protocol: Protocol,
     },
 }
 
@@ -210,7 +210,7 @@ impl TransportTuple {
     }
 
     /// Protocol.
-    pub fn protocol(&self) -> TransportProtocol {
+    pub fn protocol(&self) -> Protocol {
         let (Self::WithRemote { protocol, .. } | Self::LocalOnly { protocol, .. }) = self;
         *protocol
     }

rust/src/lib.rs

@@ -60,6 +60,7 @@ pub mod scalability_modes;
 pub mod sctp_parameters;
 pub mod srtp_parameters;
 pub mod supported_rtp_capabilities;
+pub mod webrtc_server;
 pub mod worker;
 pub mod worker_manager;