snapshot of https://github.com/pion/webrtc.git (main, 2026-09-15; blobs > 131072B omitted)

examples/rtp-forwarder/README.md

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+# 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 install github.com/pion/webrtc/v4/examples/rtp-forwarder@latest
+```
+
+### Open rtp-forwarder example page
+[jsfiddle.net](https://jsfiddle.net/fm7btvr3/) you should see your Webcam, two text-areas and `Copy browser SDP to clipboard`, `Start Session` buttons
+
+### Run rtp-forwarder, with your browsers SessionDescription as stdin
+In the jsfiddle the top textarea is your browser's Session Description. Press `Copy browser SDP to clipboard` or copy the base64 string manually.
+We will use this value in the next step.
+
+#### Linux/macOS
+Run `echo $BROWSER_SDP | rtp-forwarder`
+#### Windows
+1. Paste the SessionDescription into a file.
+1. Run `rtp-forwarder < my_file`
+
+### Input rtp-forwarder's SessionDescription into your browser
+Copy the text that `rtp-forwarder` just emitted and copy into second text area
+
+### Hit 'Start Session' in jsfiddle and enjoy your RTP forwarded stream!
+You can run any of these commands at anytime. The media is live/stateless, you can switch commands without restarting Pion.
+
+#### VLC
+Open `rtp-forwarder.sdp` with VLC and enjoy your live video!
+
+#### ffmpeg/ffprobe
+Run `ffprobe -i rtp-forwarder.sdp -protocol_whitelist file,udp,rtp` to get more details about your streams
+
+Run `ffplay -i rtp-forwarder.sdp -protocol_whitelist file,udp,rtp` to play your streams
+
+You can add `-fflags nobuffer -flags low_delay -framedrop` to lower the latency. You will have worse playback in networks with jitter. Read about minimizing the delay on [Stackoverflow](https://stackoverflow.com/a/49273163/5472819).
+
+#### Twitch/RTMP
+`ffmpeg -protocol_whitelist file,udp,rtp -i rtp-forwarder.sdp -c:v libx264 -preset veryfast -b:v 3000k -maxrate 3000k -bufsize 6000k -pix_fmt yuv420p -g 50 -c:a aac -b:a 160k -ac 2 -ar 44100 -f flv rtmp://live.twitch.tv/app/$STREAM_KEY` Make sure to replace `$STREAM_KEY` at the end of the URL first.

examples/rtp-forwarder/jsfiddle/demo.css

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+/*
+    SPDX-FileCopyrightText: 2026 The Pion community <https://pion.ly>
+    SPDX-License-Identifier: MIT
+*/
+textarea {
+    width: 500px;
+    min-height: 75px;
+}

examples/rtp-forwarder/jsfiddle/demo.details

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+---
+# SPDX-FileCopyrightText: 2026 The Pion community <https://pion.ly>
+# SPDX-License-Identifier: MIT
+
+name: rtp-forwarder
+description: Example of using Pion WebRTC to forward WebRTC streams via RTP
+authors:
+  - Quentin Renard

examples/rtp-forwarder/jsfiddle/demo.html

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+<!--
+	SPDX-FileCopyrightText: 2026 The Pion community <https://pion.ly>
+	SPDX-License-Identifier: MIT
+-->
+Browser base64 Session Description<br />
+<textarea id="localSessionDescription" readonly="true"></textarea> <br />
+<button onclick="window.copySDP()">
+	Copy browser SDP to clipboard
+</button>
+<br />
+<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

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+/* eslint-env browser */
+
+// SPDX-FileCopyrightText: 2026 The Pion community <https://pion.ly>
+// SPDX-License-Identifier: MIT
+
+const pc = new RTCPeerConnection({
+  iceServers: [
+    {
+      urls: 'stun:stun.l.google.com:19302'
+    }
+  ]
+})
+const log = msg => {
+  document.getElementById('logs').innerHTML += msg + '<br>'
+}
+
+navigator.mediaDevices.getUserMedia({ video: true, audio: true })
+  .then(stream => {
+    stream.getTracks().forEach(track => pc.addTrack(track, stream))
+    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 = () => {
+  const sd = document.getElementById('remoteSessionDescription').value
+  if (sd === '') {
+    return alert('Session Description must not be empty')
+  }
+
+  try {
+    pc.setRemoteDescription(JSON.parse(atob(sd)))
+  } catch (e) {
+    alert(e)
+  }
+}
+
+window.copySDP = () => {
+  const browserSDP = document.getElementById('localSessionDescription')
+
+  browserSDP.focus()
+  browserSDP.select()
+
+  try {
+    const successful = document.execCommand('copy')
+    const msg = successful ? 'successful' : 'unsuccessful'
+    log('Copying SDP was ' + msg)
+  } catch (err) {
+    log('Unable to copy SDP ' + err)
+  }
+}

examples/rtp-forwarder/main.go

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+// SPDX-FileCopyrightText: 2026 The Pion community <https://pion.ly>
+// SPDX-License-Identifier: MIT
+
+//go:build !js
+
+// rtp-forwarder shows how to forward your webcam/microphone via RTP using Pion WebRTC.
+package main
+
+import (
+	"bufio"
+	"encoding/base64"
+	"encoding/json"
+	"errors"
+	"fmt"
+	"io"
+	"net"
+	"os"
+	"strings"
+
+	"github.com/pion/interceptor"
+	"github.com/pion/interceptor/pkg/intervalpli"
+	"github.com/pion/rtp"
+	"github.com/pion/webrtc/v4"
+)
+
+type udpConn struct {
+	conn        *net.UDPConn
+	port        int
+	payloadType uint8
+}
+
+func main() { //nolint:gocognit,cyclop,maintidx
+	// Everything below is the Pion WebRTC API! Thanks for using it ❤️.
+
+	// Create a MediaEngine object to configure the supported codec
+	mediaEngine := &webrtc.MediaEngine{}
+
+	// Setup the codecs you want to use.
+	// We'll use a VP8 and Opus but you can also define your own
+	if err := mediaEngine.RegisterCodec(webrtc.RTPCodecParameters{
+		RTPCodecCapability: webrtc.RTPCodecCapability{
+			MimeType: webrtc.MimeTypeVP8, ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: nil,
+		},
+	}, webrtc.RTPCodecTypeVideo); err != nil {
+		panic(err)
+	}
+	if err := mediaEngine.RegisterCodec(webrtc.RTPCodecParameters{
+		RTPCodecCapability: webrtc.RTPCodecCapability{
+			MimeType: webrtc.MimeTypeOpus, ClockRate: 48000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: nil,
+		},
+	}, webrtc.RTPCodecTypeAudio); err != nil {
+		panic(err)
+	}
+
+	// Create a InterceptorRegistry. This is the user configurable RTP/RTCP Pipeline.
+	// This provides NACKs, RTCP Reports and other features. If you use `webrtc.NewPeerConnection`
+	// this is enabled by default. If you are manually managing You MUST create a InterceptorRegistry
+	// for each PeerConnection.
+	interceptorRegistry := &interceptor.Registry{}
+
+	// Register a intervalpli factory
+	// This interceptor sends a PLI every 3 seconds. A PLI causes a video keyframe to be generated by the sender.
+	// This makes our video seekable and more error resilent, but at a cost of lower picture quality and higher bitrates
+	// A real world application should process incoming RTCP packets from viewers and forward them to senders
+	intervalPliFactory, err := intervalpli.NewReceiverInterceptor()
+	if err != nil {
+		panic(err)
+	}
+	interceptorRegistry.Add(intervalPliFactory)
+
+	// Use the default set of Interceptors
+	if err = webrtc.RegisterDefaultInterceptors(mediaEngine, interceptorRegistry); err != nil {
+		panic(err)
+	}
+
+	// Create the API object with the MediaEngine
+	api := webrtc.NewAPI(webrtc.WithMediaEngine(mediaEngine), webrtc.WithInterceptorRegistry(interceptorRegistry))
+
+	// 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)
+	}
+	defer func() {
+		if cErr := peerConnection.Close(); cErr != nil {
+			fmt.Printf("cannot close peerConnection: %v\n", cErr)
+		}
+	}()
+
+	// Allow us to receive 1 audio track, and 1 video track
+	if _, err = peerConnection.AddTransceiverFromKind(webrtc.RTPCodecTypeAudio); err != nil {
+		panic(err)
+	} else if _, err = peerConnection.AddTransceiverFromKind(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
+	// Also update incoming packets with expected PayloadType, the browser may use
+	// a different value. We have to modify so our stream matches what rtp-forwarder.sdp expects
+	udpConns := map[string]*udpConn{
+		"audio": {port: 4000, payloadType: 111},
+		"video": {port: 4002, payloadType: 96},
+	}
+	for _, conn := range udpConns {
+		// Create remote addr
+		var raddr *net.UDPAddr
+		if raddr, err = net.ResolveUDPAddr("udp", fmt.Sprintf("127.0.0.1:%d", conn.port)); err != nil {
+			panic(err)
+		}
+
+		// Dial udp
+		if conn.conn, err = net.DialUDP("udp", laddr, raddr); err != nil {
+			panic(err)
+		}
+		defer func(conn net.PacketConn) {
+			if closeErr := conn.Close(); closeErr != nil {
+				panic(closeErr)
+			}
+		}(conn.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.TrackRemote, receiver *webrtc.RTPReceiver) { //nolint: revive
+		// Retrieve udp connection
+		conn, ok := udpConns[track.Kind().String()]
+		if !ok {
+			return
+		}
+
+		buf := make([]byte, 1500)
+		rtpPacket := &rtp.Packet{}
+		for {
+			// Read
+			n, _, readErr := track.Read(buf)
+			if readErr != nil {
+				panic(readErr)
+			}
+
+			// Unmarshal the packet and update the PayloadType
+			if err = rtpPacket.Unmarshal(buf[:n]); err != nil {
+				panic(err)
+			}
+			rtpPacket.PayloadType = conn.payloadType
+
+			// Marshal into original buffer with updated PayloadType
+			if n, err = rtpPacket.MarshalTo(buf); err != nil {
+				panic(err)
+			}
+
+			// Write
+			if _, writeErr := conn.conn.Write(buf[:n]); writeErr != 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
+				var opError *net.OpError
+				if errors.As(writeErr, &opError) && 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")
+		}
+	})
+
+	// Set the handler for Peer connection state
+	// This will notify you when the peer has connected/disconnected
+	peerConnection.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
+		fmt.Printf("Peer Connection State has changed: %s\n", state.String())
+
+		if state == webrtc.PeerConnectionStateFailed {
+			// Wait until PeerConnection has had no network activity for 30 seconds or another failure.
+			// It may be reconnected using an ICE Restart.
+			// Use webrtc.PeerConnectionStateDisconnected if you are interested in detecting faster timeout.
+			// Note that the PeerConnection may come back from PeerConnectionStateDisconnected.
+			fmt.Println("Done forwarding")
+			os.Exit(0)
+		}
+
+		if state == webrtc.PeerConnectionStateClosed {
+			// PeerConnection was explicitly closed. This usually happens from a DTLS CloseNotify
+			fmt.Println("Done forwarding")
+			os.Exit(0)
+		}
+	})
+
+	// Wait for the offer to be pasted
+	offer := webrtc.SessionDescription{}
+	decode(readUntilNewline(), &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)
+	}
+
+	// Create channel that is blocked until ICE Gathering is complete
+	gatherComplete := webrtc.GatheringCompletePromise(peerConnection)
+
+	// Sets the LocalDescription, and starts our UDP listeners
+	if err = peerConnection.SetLocalDescription(answer); err != nil {
+		panic(err)
+	}
+
+	// Block until ICE Gathering is complete, disabling trickle ICE
+	// we do this because we only can exchange one signaling message
+	// in a production application you should exchange ICE Candidates via OnICECandidate
+	<-gatherComplete
+
+	// Output the answer in base64 so we can paste it in browser
+	fmt.Println(encode(peerConnection.LocalDescription()))
+
+	// Block forever
+	select {}
+}
+
+// Read from stdin until we get a newline.
+func readUntilNewline() (in string) {
+	var err error
+
+	r := bufio.NewReader(os.Stdin)
+	for {
+		in, err = r.ReadString('\n')
+		if err != nil && !errors.Is(err, io.EOF) {
+			panic(err)
+		}
+
+		if in = strings.TrimSpace(in); len(in) > 0 {
+			break
+		}
+	}
+
+	fmt.Println("")
+
+	return
+}
+
+// JSON encode + base64 a SessionDescription.
+func encode(obj *webrtc.SessionDescription) string {
+	b, err := json.Marshal(obj)
+	if err != nil {
+		panic(err)
+	}
+
+	return base64.StdEncoding.EncodeToString(b)
+}
+
+// Decode a base64 and unmarshal JSON into a SessionDescription.
+func decode(in string, obj *webrtc.SessionDescription) {
+	b, err := base64.StdEncoding.DecodeString(in)
+	if err != nil {
+		panic(err)
+	}
+
+	if err = json.Unmarshal(b, obj); err != nil {
+		panic(err)
+	}
+}

examples/rtp-forwarder/rtp-forwarder.sdp

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+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

examples/rtp-forwarder/rtp-forwarder.sdp.license

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+SPDX-FileCopyrightText: 2026 The Pion community <https://pion.ly>
+SPDX-License-Identifier: MIT