From 38ee94e743ee0ff220e3ed765a194c2de1c44bc0 Mon Sep 17 00:00:00 2001 From: Quentin Renard Date: Sun, 8 Mar 2020 11:37:17 +0100 Subject: [PATCH] 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 | 1 + examples/README.md | 1 + examples/examples.json | 6 + examples/rtp-forwarder/README.md | 32 ++++ examples/rtp-forwarder/jsfiddle/demo.css | 4 + examples/rtp-forwarder/jsfiddle/demo.details | 5 + examples/rtp-forwarder/jsfiddle/demo.html | 14 ++ examples/rtp-forwarder/jsfiddle/demo.js | 38 +++++ examples/rtp-forwarder/main.go | 170 +++++++++++++++++++ examples/rtp-forwarder/rtp-forwarder.sdp | 9 + 10 files changed, 280 insertions(+) create mode 100644 examples/rtp-forwarder/README.md create mode 100644 examples/rtp-forwarder/jsfiddle/demo.css create mode 100644 examples/rtp-forwarder/jsfiddle/demo.details create mode 100644 examples/rtp-forwarder/jsfiddle/demo.html create mode 100644 examples/rtp-forwarder/jsfiddle/demo.js create mode 100644 examples/rtp-forwarder/main.go create mode 100644 examples/rtp-forwarder/rtp-forwarder.sdp diff --git a/README.md b/README.md index 0ed09dbeb72..c634e175185 100644 --- a/README.md +++ b/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 diff --git a/examples/README.md b/examples/README.md index da4113de602..eb08a27f102 100644 --- a/examples/README.md +++ b/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. diff --git a/examples/examples.json b/examples/examples.json index 1bbd6ce74af..d95d3c27686 100644 --- a/examples/examples.json +++ b/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": "#", diff --git a/examples/rtp-forwarder/README.md b/examples/rtp-forwarder/README.md new file mode 100644 index 00000000000..7e12714fd18 --- /dev/null +++ b/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 + diff --git a/examples/rtp-forwarder/jsfiddle/demo.css b/examples/rtp-forwarder/jsfiddle/demo.css new file mode 100644 index 00000000000..9e43d340755 --- /dev/null +++ b/examples/rtp-forwarder/jsfiddle/demo.css @@ -0,0 +1,4 @@ +textarea { + width: 500px; + min-height: 75px; +} \ No newline at end of file diff --git a/examples/rtp-forwarder/jsfiddle/demo.details b/examples/rtp-forwarder/jsfiddle/demo.details new file mode 100644 index 00000000000..e0b8fe3dfbf --- /dev/null +++ b/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 diff --git a/examples/rtp-forwarder/jsfiddle/demo.html b/examples/rtp-forwarder/jsfiddle/demo.html new file mode 100644 index 00000000000..cba0be079df --- /dev/null +++ b/examples/rtp-forwarder/jsfiddle/demo.html @@ -0,0 +1,14 @@ +Browser base64 Session Description
+
+ +Golang base64 Session Description
+
+
+ +
+ +Video
+
+ +Logs
+
diff --git a/examples/rtp-forwarder/jsfiddle/demo.js b/examples/rtp-forwarder/jsfiddle/demo.js new file mode 100644 index 00000000000..8599c76376a --- /dev/null +++ b/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 + '
' +} + +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) + } +} diff --git a/examples/rtp-forwarder/main.go b/examples/rtp-forwarder/main.go new file mode 100644 index 00000000000..6803c87123d --- /dev/null +++ b/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() +} diff --git a/examples/rtp-forwarder/rtp-forwarder.sdp b/examples/rtp-forwarder/rtp-forwarder.sdp new file mode 100644 index 00000000000..757f2e67b26 --- /dev/null +++ b/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 \ No newline at end of file