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

examples/rtp-to-webrtc/README.md

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+# rtp-to-webrtc
+rtp-to-webrtc demonstrates how to consume a RTP stream video UDP, and then send to a WebRTC client.
+
+With this example we have pre-made GStreamer and ffmpeg pipelines, but you can use any tool you like!
+
+## Instructions
+### Download rtp-to-webrtc
+```
+go install github.com/pion/webrtc/v4/examples/rtp-to-webrtc@latest
+```
+
+### Open jsfiddle example page
+[jsfiddle.net](https://jsfiddle.net/z7ms3u5r/) you should see two text-areas and a 'Start Session' button
+
+
+### Run rtp-to-webrtc with your browsers SessionDescription as stdin
+In the jsfiddle the top textarea is your browser's SessionDescription, copy that and:
+
+#### Linux/macOS
+Run `echo $BROWSER_SDP | rtp-to-webrtc`
+
+#### Windows
+1. Paste the SessionDescription into a file.
+1. Run `rtp-to-webrtc < my_file`
+
+### Send RTP to listening socket
+You can use any software to send VP8 packets to port 5004. We also have the pre made examples below
+
+
+#### GStreamer
+```
+gst-launch-1.0 videotestsrc ! video/x-raw,width=640,height=480,format=I420 ! vp8enc error-resilient=partitions keyframe-max-dist=10 auto-alt-ref=true cpu-used=5 deadline=1 ! rtpvp8pay ! udpsink host=127.0.0.1 port=5004
+```
+
+#### ffmpeg
+```
+ffmpeg -re -f lavfi -i testsrc=size=640x480:rate=30 -vcodec libvpx -cpu-used 5 -deadline 1 -g 10 -error-resilient 1 -auto-alt-ref 1 -f rtp 'rtp://127.0.0.1:5004?pkt_size=1200'
+```
+
+If you wish to send audio replace all occurrences of `vp8` with Opus in `main.go` then run
+
+```
+ffmpeg -f lavfi -i 'sine=frequency=1000' -c:a libopus -b:a 48000 -sample_fmt s16p -ssrc 1 -payload_type 111 -f rtp -max_delay 0 -application lowdelay 'rtp://127.0.0.1:5004?pkt_size=1200'
+```
+
+If you wish to send H264 instead of VP8 replace all occurrences of `vp8` with H264 in `main.go` then run
+
+```
+ffmpeg -re -f lavfi -i testsrc=size=640x480:rate=30 -pix_fmt yuv420p -c:v libx264 -g 10 -preset ultrafast -tune zerolatency -f rtp 'rtp://127.0.0.1:5004?pkt_size=1200'
+```
+
+### Input rtp-to-webrtc's SessionDescription into your browser
+Copy the text that `rtp-to-webrtc` just emitted and copy into second text area
+
+### Hit 'Start Session' in jsfiddle, enjoy your video!
+A video should start playing in your browser above the input boxes.
+
+Congrats, you have used Pion WebRTC! Now start building something cool
+
+## Dealing with broken/lossy inputs
+Pion WebRTC also provides a [SampleBuilder](https://pkg.go.dev/github.com/pion/webrtc/v3@v3.0.4/pkg/media/samplebuilder). This consumes RTP packets and returns samples.
+It can be used to re-order and delay for lossy streams. You can see its usage in this example in [daf27b](https://github.com/pion/webrtc/commit/daf27bd0598233b57428b7809587ec3c09510413).
+
+Currently it isn't working with H264, but is useful for VP8 and Opus. See [#1652](https://github.com/pion/webrtc/issues/1652) for the status of fixing for H264.

examples/rtp-to-webrtc/main.go

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+// SPDX-FileCopyrightText: 2026 The Pion community <https://pion.ly>
+// SPDX-License-Identifier: MIT
+
+//go:build !js
+
+// rtp-to-webrtc demonstrates how to consume a RTP stream video UDP, and then send to a WebRTC client.
+package main
+
+import (
+	"bufio"
+	"encoding/base64"
+	"encoding/json"
+	"errors"
+	"fmt"
+	"io"
+	"net"
+	"os"
+	"strings"
+
+	"github.com/pion/webrtc/v4"
+)
+
+// nolint:cyclop
+func main() {
+	peerConnection, err := webrtc.NewPeerConnection(webrtc.Configuration{
+		ICEServers: []webrtc.ICEServer{
+			{
+				URLs: []string{"stun:stun.l.google.com:19302"},
+			},
+		},
+	})
+	if err != nil {
+		panic(err)
+	}
+
+	// Open a UDP Listener for RTP Packets on port 5004
+	listener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: 5004})
+	if err != nil {
+		panic(err)
+	}
+
+	// Increase the UDP receive buffer size
+	// Default UDP buffer sizes vary on different operating systems
+	bufferSize := 300000 // 300KB
+	err = listener.SetReadBuffer(bufferSize)
+	if err != nil {
+		panic(err)
+	}
+
+	defer func() {
+		if err = listener.Close(); err != nil {
+			panic(err)
+		}
+	}()
+
+	// Create a video track
+	videoTrack, err := webrtc.NewTrackLocalStaticRTP(
+		webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeVP8}, "video", "pion",
+	)
+	if err != nil {
+		panic(err)
+	}
+	rtpSender, err := peerConnection.AddTrack(videoTrack)
+	if 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.
+	go func() {
+		rtcpBuf := make([]byte, 1500)
+		for {
+			if _, _, rtcpErr := rtpSender.Read(rtcpBuf); rtcpErr != nil {
+				return
+			}
+		}
+	}()
+
+	// 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.ICEConnectionStateFailed {
+			if closeErr := peerConnection.Close(); closeErr != nil {
+				panic(closeErr)
+			}
+		}
+	})
+
+	// 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()))
+
+	// Read RTP packets forever and send them to the WebRTC Client
+	inboundRTPPacket := make([]byte, 1600) // UDP MTU
+	for {
+		n, _, err := listener.ReadFrom(inboundRTPPacket)
+		if err != nil {
+			panic(fmt.Sprintf("error during read: %s", err))
+		}
+
+		if _, err = videoTrack.Write(inboundRTPPacket[:n]); err != nil {
+			if errors.Is(err, io.ErrClosedPipe) {
+				// The peerConnection has been closed.
+				return
+			}
+
+			panic(err)
+		}
+	}
+}
+
+// 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)
+	}
+}