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

examples/simulcast/README.md

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+# simulcast
+demonstrates of how to handle incoming track with multiple simulcast rtp streams and show all them back.
+
+The browser will not send higher quality streams unless it has the available bandwidth. You can look at
+the bandwidth estimation in `chrome://webrtc-internals`. It is under `VideoBwe` when `Read Stats From: Legacy non-Standard`
+is selected.
+
+## Instructions
+### Download simulcast
+```
+go install github.com/pion/webrtc/v4/examples/simulcast@latest
+```
+
+### Open simulcast example page
+[jsfiddle.net](https://jsfiddle.net/tz4d5bhj/) you should see two text-areas and two buttons: `Copy browser SDP to clipboard`, `Start Session`.
+
+### Run simulcast, 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 | simulcast`
+#### Windows
+1. Paste the SessionDescription into a file.
+1. Run `simulcast < my_file`
+
+### Input simulcast's SessionDescription into your browser
+Copy the text that `simulcast` just emitted and copy into second text area
+
+### Hit 'Start Session' in jsfiddle, enjoy your video!
+Your browser should send a simulcast track to Pion, and then all 3 incoming streams will be relayed back.
+
+Congrats, you have used Pion WebRTC! Now start building something cool

examples/simulcast/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/simulcast/jsfiddle/demo.details

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+---
+# SPDX-FileCopyrightText: 2026 The Pion community <https://pion.ly>
+# SPDX-License-Identifier: MIT
+
+name: simulcast
+description: Example of how to have Pion handle incoming track with multiple simulcast rtp streams and show all them back.
+authors:
+  - Simone Gotti

examples/simulcast/jsfiddle/demo.html

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+<!--
+	SPDX-FileCopyrightText: 2026 The Pion community <https://pion.ly>
+	SPDX-License-Identifier: MIT
+-->
+<script src="https://webrtc.github.io/adapter/adapter-latest.js"></script>
+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 />
+
+<div>
+  Browser stream<br />
+  <video id="browserVideo" width="200" height="200" autoplay muted></video>
+</div>
+
+<div id="serverVideos">
+  Video from server<br />
+</div>

examples/simulcast/jsfiddle/demo.js

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+/* eslint-env browser */
+
+// SPDX-FileCopyrightText: 2026 The Pion community <https://pion.ly>
+// SPDX-License-Identifier: MIT
+
+// Create peer conn
+const pc = new RTCPeerConnection({
+  iceServers: [{
+    urls: 'stun:stun.l.google.com:19302'
+  }]
+})
+
+pc.oniceconnectionstatechange = (e) => {
+  console.log('connection state change', pc.iceConnectionState)
+}
+pc.onicecandidate = (event) => {
+  if (event.candidate === null) {
+    document.getElementById('localSessionDescription').value = btoa(
+      JSON.stringify(pc.localDescription)
+    )
+  }
+}
+
+pc.onnegotiationneeded = (e) =>
+  pc
+    .createOffer()
+    .then((d) => pc.setLocalDescription(d))
+    .catch(console.error)
+
+pc.ontrack = (event) => {
+  console.log('Got track event', event)
+  const video = document.createElement('video')
+  video.srcObject = event.streams[0]
+  video.autoplay = true
+  video.width = '500'
+  const label = document.createElement('div')
+  label.textContent = event.streams[0].id
+  document.getElementById('serverVideos').appendChild(label)
+  document.getElementById('serverVideos').appendChild(video)
+}
+
+navigator.mediaDevices
+  .getUserMedia({
+    video: {
+      width: {
+        ideal: 4096
+      },
+      height: {
+        ideal: 2160
+      },
+      frameRate: {
+        ideal: 60,
+        min: 10
+      }
+    },
+    audio: false
+  })
+  .then((stream) => {
+    document.getElementById('browserVideo').srcObject = stream
+    pc.addTransceiver(stream.getVideoTracks()[0], {
+      direction: 'sendonly',
+      streams: [stream],
+      sendEncodings: [
+        // for firefox order matters... first high resolution, then scaled resolutions...
+        {
+          rid: 'f'
+        },
+        {
+          rid: 'h',
+          scaleResolutionDownBy: 2.0
+        },
+        {
+          rid: 'q',
+          scaleResolutionDownBy: 4.0
+        }
+      ]
+    })
+    pc.addTransceiver('video')
+    pc.addTransceiver('video')
+    pc.addTransceiver('video')
+  })
+
+window.startSession = () => {
+  const sd = document.getElementById('remoteSessionDescription').value
+  if (sd === '') {
+    return alert('Session Description must not be empty')
+  }
+
+  try {
+    console.log('answer', JSON.parse(atob(sd)))
+    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'
+    console.log('Copying SDP was ' + msg)
+  } catch (err) {
+    console.log('Unable to copy SDP ' + err)
+  }
+}

examples/simulcast/main.go

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+// SPDX-FileCopyrightText: 2026 The Pion community <https://pion.ly>
+// SPDX-License-Identifier: MIT
+
+//go:build !js
+
+// simulcast demonstrates of how to handle incoming track with multiple simulcast rtp streams and show all them back.
+package main
+
+import (
+	"bufio"
+	"encoding/base64"
+	"encoding/json"
+	"errors"
+	"fmt"
+	"io"
+	"os"
+	"strings"
+	"time"
+
+	"github.com/pion/rtcp"
+	"github.com/pion/webrtc/v4"
+)
+
+// nolint:gocognit, cyclop
+func main() {
+	// 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 := webrtc.NewPeerConnection(config)
+	if err != nil {
+		panic(err)
+	}
+	defer func() {
+		if cErr := peerConnection.Close(); cErr != nil {
+			fmt.Printf("cannot close peerConnection: %v\n", cErr)
+		}
+	}()
+
+	outputTracks := map[string]*webrtc.TrackLocalStaticRTP{}
+
+	// Create Track that we send video back to browser on
+	outputTrack, err := webrtc.NewTrackLocalStaticRTP(webrtc.RTPCodecCapability{
+		MimeType: webrtc.MimeTypeVP8,
+	}, "video_q", "pion_q")
+	if err != nil {
+		panic(err)
+	}
+	outputTracks["q"] = outputTrack
+
+	outputTrack, err = webrtc.NewTrackLocalStaticRTP(webrtc.RTPCodecCapability{
+		MimeType: webrtc.MimeTypeVP8,
+	}, "video_h", "pion_h")
+	if err != nil {
+		panic(err)
+	}
+	outputTracks["h"] = outputTrack
+
+	outputTrack, err = webrtc.NewTrackLocalStaticRTP(webrtc.RTPCodecCapability{
+		MimeType: webrtc.MimeTypeVP8,
+	}, "video_f", "pion_f")
+	if err != nil {
+		panic(err)
+	}
+	outputTracks["f"] = outputTrack
+
+	if _, err = peerConnection.AddTransceiverFromKind(
+		webrtc.RTPCodecTypeVideo,
+		webrtc.RTPTransceiverInit{Direction: webrtc.RTPTransceiverDirectionRecvonly},
+	); err != nil {
+		panic(err)
+	}
+
+	// Add this newly created track to the PeerConnection to send back video
+	if _, err = peerConnection.AddTransceiverFromTrack(
+		outputTracks["q"], webrtc.RTPTransceiverInit{Direction: webrtc.RTPTransceiverDirectionSendonly}); err != nil {
+		panic(err)
+	}
+	if _, err = peerConnection.AddTransceiverFromTrack(
+		outputTracks["h"],
+		webrtc.RTPTransceiverInit{Direction: webrtc.RTPTransceiverDirectionSendonly},
+	); err != nil {
+		panic(err)
+	}
+	if _, err = peerConnection.AddTransceiverFromTrack(
+		outputTracks["f"],
+		webrtc.RTPTransceiverInit{Direction: webrtc.RTPTransceiverDirectionSendonly},
+	); err != nil {
+		panic(err)
+	}
+
+	// Read incoming RTCP packets
+	// Before these packets are returned they are processed by interceptors. For things
+	// like NACK this needs to be called.
+	processRTCP := func(rtpSender *webrtc.RTPSender) {
+		rtcpBuf := make([]byte, 1500)
+		for {
+			if _, _, rtcpErr := rtpSender.Read(rtcpBuf); rtcpErr != nil {
+				return
+			}
+		}
+	}
+	for _, rtpSender := range peerConnection.GetSenders() {
+		go processRTCP(rtpSender)
+	}
+
+	// Wait for the offer to be pasted
+	offer := webrtc.SessionDescription{}
+	decode(readUntilNewline(), &offer)
+
+	if err = peerConnection.SetRemoteDescription(offer); err != nil {
+		panic(err)
+	}
+
+	// Set a handler for when a new remote track starts
+	peerConnection.OnTrack(func(track *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) { //nolint: revive
+		fmt.Println("Track has started")
+
+		// Start reading from all the streams and sending them to the related output track
+		rid := track.RID()
+		if track.Kind() == webrtc.RTPCodecTypeVideo {
+			go func() {
+				ticker := time.NewTicker(3 * time.Second)
+				defer ticker.Stop()
+				for range ticker.C {
+					fmt.Printf("Sending pli for stream with rid: %q, ssrc: %d\n", track.RID(), track.SSRC())
+					if writeErr := peerConnection.WriteRTCP(
+						[]rtcp.Packet{&rtcp.PictureLossIndication{MediaSSRC: uint32(track.SSRC())}},
+					); writeErr != nil {
+						fmt.Println(writeErr)
+					}
+				}
+			}()
+		}
+		for {
+			// Read RTP packets being sent to Pion
+			packet, _, readErr := track.ReadRTP()
+			if readErr != nil {
+				panic(readErr)
+			}
+
+			if writeErr := outputTracks[rid].WriteRTP(packet); writeErr != nil && !errors.Is(writeErr, io.ErrClosedPipe) {
+				panic(writeErr)
+			}
+		}
+	})
+
+	// 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("Peer Connection has gone to failed exiting")
+			os.Exit(0)
+		}
+
+		if state == webrtc.PeerConnectionStateClosed {
+			// PeerConnection was explicitly closed. This usually happens from a DTLS CloseNotify
+			fmt.Println("Peer Connection has gone to closed exiting")
+			os.Exit(0)
+		}
+	})
+
+	// Create an 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
+	err = peerConnection.SetLocalDescription(answer)
+	if 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)
+	}
+}