diff --git a/README.md b/README.md
index 0ed09dbe..c634e175 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 da4113de..eb08a27f 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 1bbd6ce7..d95d3c27 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 00000000..7e12714f
--- /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 00000000..9e43d340
--- /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 00000000..e0b8fe3d
--- /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 00000000..cba0be07
--- /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 00000000..8599c763
--- /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 00000000..6803c871
--- /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 00000000..757f2e67
--- /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