Added examples/rtp-forwarder

Add new example that demonstrates how to take WebRTC to RTP.
Also provides instructions and pre-canned SDP so you can easily
playback in VLC and ffmpeg.

Resolves #1061

README.md

@@ -141,6 +141,7 @@ Check out the **[contributing wiki](https://github.com/pion/webrtc/wiki/Contribu
 * [lawl](https://github.com/lawl)
 * [Jorropo](https://github.com/Jorropo)
 * [Akil](https://github.com/akilude)
+* [Quentin Renard](https://github.com/asticode)
 
 ### License
 MIT License - see [LICENSE](LICENSE) for full text

examples/README.md

@@ -12,6 +12,7 @@ For more full featured examples that use 3rd party libraries see our **[example-
 * [Play from disk](play-from-disk): The play-from-disk example demonstrates how to send video to your browser from a file saved to disk.
 * [Save to Disk](save-to-disk): The save-to-disk example shows how to record your webcam and save the footage to disk on the server side.
 * [Broadcast](broadcast): The broadcast example demonstrates how to broadcast a video to multiple peers. A broadcaster uploads the video once and the server forwards it to all other peers.
+* [RTP Forwarder](rtp-forwarder): The rtp-forwarder example demonstrates how to forward your audio/video streams using RTP.
 
 #### Data Channel API
 * [Data Channels](data-channels): The data-channels example shows how you can send/recv DataChannel messages from a web browser.

examples/examples.json

@@ -53,6 +53,12 @@
 		"description": "The broadcast example demonstrates how to broadcast a video to multiple peers. A broadcaster uploads the video once and the server forwards it to all other peers.",
 		"type": "browser"
 	},
+	{
+		"title": "RTP Forwarder",
+		"link": "rtp-forwarder",
+		"description": "The rtp-forwarder example demonstrates how to forward your audio/video streams using RTP.",
+		"type": "browser"
+	},
 	{
 		"title": "Custom Logger",
 		"link": "#",

examples/rtp-forwarder/README.md

@@ -0,0 +1,32 @@
+# rtp-forwarder
+rtp-forwarder is a simple application that shows how to forward your webcam/microphone via RTP using Pion WebRTC.
+
+## Instructions
+### Download rtp-forwarder
+```
+go get github.com/pion/webrtc/examples/rtp-forwarder
+```
+
+### Open rtp-forwarder example page
+[jsfiddle.net](https://jsfiddle.net/sq69370h/) you should see your Webcam, two text-areas and a 'Start Session' button
+
+### Run rtp-forwarder, with your browsers SessionDescription as stdin
+In the jsfiddle the top textarea is your browser, copy that and:
+#### Linux/macOS
+Run `echo $BROWSER_SDP | rtp-forwarder`
+#### Windows
+1. Paste the SessionDescription into a file.
+1. Run `rtp-forwarder < my_file`
+
+### Input rtp-forwarder's SessionDescription into your browser
+Copy the text that `rtp-forwarder` just emitted and copy into second text area
+
+### Hit 'Start Session' in jsfiddle and enjoy your RTP forwarded stream!
+#### VLC
+Open `rtp-forwarder.sdp` with VLC and enjoy your live video!
+
+### ffmpeg/ffprobe
+Run `ffprobe -i rtp-forwarder.sdp -protocol_whitelist file,udp,rtp` to get more details about your streams
+
+Run `ffplay -i rtp-forwarder.sdp -protocol_whitelist file,udp,rtp` to play your streams
+

examples/rtp-forwarder/jsfiddle/demo.css

@@ -0,0 +1,4 @@
+textarea {
+    width: 500px;
+    min-height: 75px;
+}

examples/rtp-forwarder/jsfiddle/demo.details

@@ -0,0 +1,5 @@
+---
+ name: rtp-forwarder
+ description: Example of using Pion WebRTC to forward WebRTC streams via RTP
+ authors:
+   - Quentin Renard

examples/rtp-forwarder/jsfiddle/demo.html

@@ -0,0 +1,14 @@
+Browser base64 Session Description<br />
+<textarea id="localSessionDescription" readonly="true"></textarea> <br />
+
+Golang base64 Session Description<br />
+<textarea id="remoteSessionDescription"></textarea> <br/>
+<button onclick="window.startSession()"> Start Session </button><br />
+
+<br />
+
+Video<br />
+<video id="video1" width="160" height="120" autoplay muted></video> <br />
+
+Logs<br />
+<div id="logs"></div>

examples/rtp-forwarder/jsfiddle/demo.js

@@ -0,0 +1,38 @@
+/* eslint-env browser */
+
+let pc = new RTCPeerConnection({
+  iceServers: [
+    {
+      urls: 'stun:stun.l.google.com:19302'
+    }
+  ]
+})
+var log = msg => {
+  document.getElementById('logs').innerHTML += msg + '<br>'
+}
+
+navigator.mediaDevices.getUserMedia({ video: true, audio: true })
+  .then(stream => {
+    pc.addStream(document.getElementById('video1').srcObject = stream)
+    pc.createOffer().then(d => pc.setLocalDescription(d)).catch(log)
+  }).catch(log)
+
+pc.oniceconnectionstatechange = e => log(pc.iceConnectionState)
+pc.onicecandidate = event => {
+  if (event.candidate === null) {
+    document.getElementById('localSessionDescription').value = btoa(JSON.stringify(pc.localDescription))
+  }
+}
+
+window.startSession = () => {
+  let sd = document.getElementById('remoteSessionDescription').value
+  if (sd === '') {
+    return alert('Session Description must not be empty')
+  }
+
+  try {
+    pc.setRemoteDescription(new RTCSessionDescription(JSON.parse(atob(sd))))
+  } catch (e) {
+    alert(e)
+  }
+}

examples/rtp-forwarder/main.go

@@ -0,0 +1,170 @@
+package main
+
+import (
+	"context"
+	"fmt"
+	"net"
+	"time"
+
+	"github.com/pion/rtcp"
+	"github.com/pion/webrtc/v2"
+	"github.com/pion/webrtc/v2/examples/internal/signal"
+)
+
+type udpConn struct {
+	conn *net.UDPConn
+	port int
+}
+
+func main() {
+	// Create context
+	ctx, cancel := context.WithCancel(context.Background())
+
+	// Create a MediaEngine object to configure the supported codec
+	m := webrtc.MediaEngine{}
+
+	// Setup the codecs you want to use.
+	// We'll use a VP8 codec but you can also define your own
+	m.RegisterCodec(webrtc.NewRTPOpusCodec(webrtc.DefaultPayloadTypeOpus, 48000))
+	m.RegisterCodec(webrtc.NewRTPVP8Codec(webrtc.DefaultPayloadTypeVP8, 90000))
+
+	// Create the API object with the MediaEngine
+	api := webrtc.NewAPI(webrtc.WithMediaEngine(m))
+
+	// Everything below is the Pion WebRTC API! Thanks for using it ❤️.
+
+	// Prepare the configuration
+	config := webrtc.Configuration{
+		ICEServers: []webrtc.ICEServer{
+			{
+				URLs: []string{"stun:stun.l.google.com:19302"},
+			},
+		},
+	}
+
+	// Create a new RTCPeerConnection
+	peerConnection, err := api.NewPeerConnection(config)
+	if err != nil {
+		panic(err)
+	}
+
+	// Allow us to receive 1 audio track, and 1 video track
+	if _, err = peerConnection.AddTransceiver(webrtc.RTPCodecTypeAudio); err != nil {
+		panic(err)
+	} else if _, err = peerConnection.AddTransceiver(webrtc.RTPCodecTypeVideo); err != nil {
+		panic(err)
+	}
+
+	// Create a local addr
+	var laddr *net.UDPAddr
+	if laddr, err = net.ResolveUDPAddr("udp", "127.0.0.1:"); err != nil {
+		panic(err)
+	}
+
+	// Prepare udp conns
+	udpConns := map[string]*udpConn{
+		"audio": {port: 4000},
+		"video": {port: 4002},
+	}
+	for _, c := range udpConns {
+		// Create remote addr
+		var raddr *net.UDPAddr
+		if raddr, err = net.ResolveUDPAddr("udp", fmt.Sprintf("127.0.0.1:%d", c.port)); err != nil {
+			panic(err)
+		}
+
+		// Dial udp
+		if c.conn, err = net.DialUDP("udp", laddr, raddr); err != nil {
+			panic(err)
+		}
+		defer func(conn net.PacketConn) {
+			if closeErr := conn.Close(); closeErr != nil {
+				panic(closeErr)
+			}
+		}(c.conn)
+	}
+
+	// Set a handler for when a new remote track starts, this handler will forward data to
+	// our UDP listeners.
+	// In your application this is where you would handle/process audio/video
+	peerConnection.OnTrack(func(track *webrtc.Track, receiver *webrtc.RTPReceiver) {
+		// Retrieve udp connection
+		c, ok := udpConns[track.Kind().String()]
+		if !ok {
+			return
+		}
+
+		// Send a PLI on an interval so that the publisher is pushing a keyframe every rtcpPLIInterval
+		go func() {
+			ticker := time.NewTicker(time.Second * 2)
+			for range ticker.C {
+				if rtcpErr := peerConnection.WriteRTCP([]rtcp.Packet{&rtcp.PictureLossIndication{MediaSSRC: track.SSRC()}}); rtcpErr != nil {
+					fmt.Println(rtcpErr)
+				}
+			}
+		}()
+
+		b := make([]byte, 1500)
+		for {
+			// Read
+			n, readErr := track.Read(b)
+			if readErr != nil {
+				panic(readErr)
+			}
+
+			// Write
+			if _, err = c.conn.Write(b[:n]); err != nil {
+				// For this particular example, third party applications usually timeout after a short
+				// amount of time during which the user doesn't have enough time to provide the answer
+				// to the browser.
+				// That's why, for this particular example, the user first needs to provide the answer
+				// to the browser then open the third party application. Therefore we must not kill
+				// the forward on "connection refused" errors
+				if opError, ok := err.(*net.OpError); ok && opError.Err.Error() == "write: connection refused" {
+					continue
+				}
+				panic(err)
+			}
+		}
+	})
+
+	// Set the handler for ICE connection state
+	// This will notify you when the peer has connected/disconnected
+	peerConnection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
+		fmt.Printf("Connection State has changed %s \n", connectionState.String())
+
+		if connectionState == webrtc.ICEConnectionStateConnected {
+			fmt.Println("Ctrl+C the remote client to stop the demo")
+		} else if connectionState == webrtc.ICEConnectionStateFailed ||
+			connectionState == webrtc.ICEConnectionStateDisconnected {
+			fmt.Println("Done forwarding")
+			cancel()
+		}
+	})
+
+	// Wait for the offer to be pasted
+	offer := webrtc.SessionDescription{}
+	signal.Decode(signal.MustReadStdin(), &offer)
+
+	// Set the remote SessionDescription
+	if err = peerConnection.SetRemoteDescription(offer); err != nil {
+		panic(err)
+	}
+
+	// Create answer
+	answer, err := peerConnection.CreateAnswer(nil)
+	if err != nil {
+		panic(err)
+	}
+
+	// Sets the LocalDescription, and starts our UDP listeners
+	if err = peerConnection.SetLocalDescription(answer); err != nil {
+		panic(err)
+	}
+
+	// Output the answer in base64 so we can paste it in browser
+	fmt.Println(signal.Encode(answer))
+
+	// Wait for context to be done
+	<-ctx.Done()
+}

examples/rtp-forwarder/rtp-forwarder.sdp

@@ -0,0 +1,9 @@
+v=0
+o=- 0 0 IN IP4 127.0.0.1
+s=Pion WebRTC
+c=IN IP4 127.0.0.1
+t=0 0
+m=audio 4000 RTP/AVP 111
+a=rtpmap:111 OPUS/48000/2
+m=video 4002 RTP/AVP 96
+a=rtpmap:96 VP8/90000