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

examples/whip-whep/README.md

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+# whip-whep
+whip-whep demonstrates using WHIP and WHEP with Pion. Since WHIP+WHEP is standardized signaling you can publish via tools like OBS and GStreamer.
+You can then watch it in sub-second time from your browser, or pull the video back into OBS and GStreamer via WHEP.
+
+Further details about the why and how of WHIP+WHEP are below the instructions.
+
+## Instructions
+
+### Download whip-whep
+
+This example requires you to clone the repo since it is serving static HTML.
+
+```
+git clone https://github.com/pion/webrtc.git
+cd webrtc/examples/whip-whep
+```
+
+### Run whip-whep
+Execute `go run *.go`
+
+### Publish
+
+You can publish via an tool that supports WHIP or via your browser. To publish via your browser open [http://localhost:8080](http://localhost:8080), and press publish.
+
+To publish via OBS set `Service` to `WHIP` and `Server` to `http://localhost:8080/whip`. The `Bearer Token` can be whatever value you like.
+
+
+### Subscribe
+
+Once you have started publishing open [http://localhost:8080](http://localhost:8080) and press the subscribe button. You can now view your video you published via
+OBS or your browser.
+
+Congrats, you have used Pion WebRTC! Now start building something cool
+
+## Why WHIP/WHEP?
+
+WHIP/WHEP mandates that a Offer is uploaded via HTTP. The server responds with a Answer. With this strong API contract WebRTC support can be added to tools like OBS.
+
+For more info on WHIP/WHEP specification, feel free to read some of these great resources:
+- https://webrtchacks.com/webrtc-cracks-the-whip-on-obs/
+- https://datatracker.ietf.org/doc/draft-ietf-wish-whip/
+- https://datatracker.ietf.org/doc/draft-ietf-wish-whep/
+- https://bloggeek.me/whip-whep-webrtc-live-streaming

examples/whip-whep/index.html

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+<html>
+
+  <!--
+		SPDX-FileCopyrightText: 2026 The Pion community <https://pion.ly>
+		SPDX-License-Identifier: MIT
+	-->
+  <head>
+    <title>whip-whep</title>
+  </head>
+
+  <body>
+    <button onclick="window.doWHIP()">Publish</button>
+    <button onclick="window.doWHEP()">Subscribe</button>
+    <h3> Video </h3>
+    <video id="videoPlayer" autoplay muted controls style="width: 500"> </video>
+
+
+    <h3> ICE Connection States </h3>
+    <div id="iceConnectionStates"></div> <br />
+  </body>
+
+  <script>
+    let peerConnection = new RTCPeerConnection()
+
+    peerConnection.oniceconnectionstatechange = () => {
+      let el = document.createElement('p')
+      el.appendChild(document.createTextNode(peerConnection.iceConnectionState))
+
+      document.getElementById('iceConnectionStates').appendChild(el);
+    }
+
+    window.doWHEP = () => {
+      peerConnection.addTransceiver('video', { direction: 'recvonly' })
+      peerConnection.addTransceiver('audio', { direction: 'recvonly' })
+
+      peerConnection.ontrack = function (event) {
+        document.getElementById('videoPlayer').srcObject = event.streams[0]
+      }
+
+      peerConnection.createOffer().then(offer => {
+        peerConnection.setLocalDescription(offer)
+
+        fetch(`/whep`, {
+          method: 'POST',
+          body: offer.sdp,
+          headers: {
+            Authorization: `Bearer none`,
+            'Content-Type': 'application/sdp'
+          }
+        }).then(r => r.text())
+          .then(answer => {
+            peerConnection.setRemoteDescription({
+              sdp: answer,
+              type: 'answer'
+            })
+          })
+      })
+    }
+
+    window.doWHIP = () => {
+      navigator.mediaDevices.getUserMedia({ video: true, audio: true })
+        .then(stream => {
+          document.getElementById('videoPlayer').srcObject = stream
+          stream.getTracks().forEach(track => peerConnection.addTrack(track, stream))
+
+          peerConnection.createOffer().then(offer => {
+            peerConnection.setLocalDescription(offer)
+
+            fetch(`/whip`, {
+              method: 'POST',
+              body: offer.sdp,
+              headers: {
+                Authorization: `Bearer none`,
+                'Content-Type': 'application/sdp'
+              }
+            }).then(r => r.text())
+              .then(answer => {
+                peerConnection.setRemoteDescription({
+                  sdp: answer,
+                  type: 'answer'
+                })
+              })
+          })
+      })
+    }
+  </script>
+</html>

examples/whip-whep/main.go

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+// SPDX-FileCopyrightText: 2026 The Pion community <https://pion.ly>
+// SPDX-License-Identifier: MIT
+
+//go:build !js
+
+// whip-whep demonstrates how to use the WHIP/WHEP specifications to exchange SPD descriptions
+// and stream media to a WebRTC client in the browser or OBS.
+package main
+
+import (
+	"errors"
+	"fmt"
+	"io"
+	"net/http"
+
+	"github.com/pion/interceptor"
+	"github.com/pion/interceptor/pkg/intervalpli"
+	"github.com/pion/webrtc/v4"
+)
+
+// nolint: gochecknoglobals
+var (
+	videoTrack *webrtc.TrackLocalStaticRTP
+	audioTrack *webrtc.TrackLocalStaticRTP
+
+	peerConnectionConfiguration = webrtc.Configuration{
+		ICEServers: []webrtc.ICEServer{
+			{
+				URLs: []string{"stun:stun.l.google.com:19302"},
+			},
+		},
+	}
+)
+
+// nolint:gocognit
+func main() {
+	// Everything below is the Pion WebRTC API! Thanks for using it ❤️.
+	var err error
+	if videoTrack, err = webrtc.NewTrackLocalStaticRTP(webrtc.RTPCodecCapability{
+		MimeType: webrtc.MimeTypeH264,
+	}, "video", "pion"); err != nil {
+		panic(err)
+	}
+	if audioTrack, err = webrtc.NewTrackLocalStaticRTP(webrtc.RTPCodecCapability{
+		MimeType: webrtc.MimeTypeOpus,
+	}, "audio", "pion"); err != nil {
+		panic(err)
+	}
+
+	http.Handle("/", http.FileServer(http.Dir(".")))
+	http.HandleFunc("/whep", whepHandler)
+	http.HandleFunc("/whip", whipHandler)
+
+	fmt.Println("Open http://localhost:8080 to access this demo")
+	panic(http.ListenAndServe(":8080", nil)) // nolint: gosec
+}
+
+func whipHandler(res http.ResponseWriter, req *http.Request) { // nolint: cyclop
+	fmt.Printf("Request to %s, method = %s\n", req.URL, req.Method)
+
+	res.Header().Add("Access-Control-Allow-Origin", "*")
+	res.Header().Add("Access-Control-Allow-Methods", "POST")
+	res.Header().Add("Access-Control-Allow-Headers", "*")
+	res.Header().Add("Access-Control-Allow-Headers", "Authorization")
+
+	if req.Method == http.MethodOptions {
+		return
+	}
+
+	// Read the offer from HTTP Request
+	offer, err := io.ReadAll(req.Body)
+	if err != nil {
+		panic(err)
+	}
+
+	// Create a MediaEngine object to configure the supported codec
+	mediaEngine := &webrtc.MediaEngine{}
+
+	// Set up the codecs you want to use.
+	// We'll only use H264 and Opus but you can also define your own
+	if err = mediaEngine.RegisterCodec(webrtc.RTPCodecParameters{
+		RTPCodecCapability: webrtc.RTPCodecCapability{
+			MimeType: webrtc.MimeTypeH264, ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: nil,
+		},
+		PayloadType: 96,
+	}, webrtc.RTPCodecTypeVideo); err != nil {
+		panic(err)
+	}
+	if err = mediaEngine.RegisterCodec(webrtc.RTPCodecParameters{
+		RTPCodecCapability: webrtc.RTPCodecCapability{
+			MimeType: webrtc.MimeTypeOpus, ClockRate: 48000, Channels: 2, SDPFmtpLine: "", RTCPFeedback: nil,
+		},
+		PayloadType: 97,
+	}, webrtc.RTPCodecTypeAudio); err != nil {
+		panic(err)
+	}
+
+	// Create an 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))
+
+	// Create a new RTCPeerConnection
+	peerConnection, err := api.NewPeerConnection(peerConnectionConfiguration)
+	if err != nil {
+		panic(err)
+	}
+
+	// Allow us to receive 1 video track and 1 audio track
+	if _, err = peerConnection.AddTransceiverFromKind(webrtc.RTPCodecTypeVideo); err != nil {
+		panic(err)
+	}
+	if _, err = peerConnection.AddTransceiverFromKind(webrtc.RTPCodecTypeAudio); err != nil {
+		panic(err)
+	}
+
+	// Set a handler for when a new remote track starts, this handler saves buffers to disk as
+	// an ivf file, since we could have multiple video tracks we provide a counter.
+	// In your application this is where you would handle/process video
+	peerConnection.OnTrack(func(track *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) {
+		go func() {
+			for {
+				_, _, err := receiver.ReadRTCP()
+				if err != nil {
+					if errors.Is(err, io.EOF) {
+						fmt.Printf("***** EOF reading RTCP from publish peer connection\n")
+
+						break
+					}
+					panic(err)
+				}
+			}
+		}()
+		go func() {
+			for {
+				pkt, _, err := track.ReadRTP()
+				if err != nil {
+					if errors.Is(err, io.EOF) {
+						fmt.Printf("***** EOF reading RTP from publish peer connection\n")
+
+						break
+					}
+					panic(err)
+				}
+
+				// Strip any WHIP extensions before forwarding to WHEP
+				pkt.Header.Extensions = nil
+				pkt.Header.Extension = false
+
+				if track.Kind() == webrtc.RTPCodecTypeVideo {
+					if err = videoTrack.WriteRTP(pkt); err != nil {
+						panic(err)
+					}
+				} else if track.Kind() == webrtc.RTPCodecTypeAudio {
+					if err = audioTrack.WriteRTP(pkt); err != nil {
+						panic(err)
+					}
+				}
+			}
+		}()
+	})
+	// Send answer via HTTP Response
+	writeAnswer(res, peerConnection, offer, "/whip")
+}
+
+func whepHandler(res http.ResponseWriter, req *http.Request) { //nolint:cyclop
+	fmt.Printf("Request to %s, method = %s\n", req.URL, req.Method)
+
+	res.Header().Add("Access-Control-Allow-Origin", "*")
+	res.Header().Add("Access-Control-Allow-Methods", "POST")
+	res.Header().Add("Access-Control-Allow-Headers", "*")
+	res.Header().Add("Access-Control-Allow-Headers", "Authorization")
+
+	if req.Method == http.MethodOptions {
+		return
+	}
+
+	// Read the offer from HTTP Request
+	offer, err := io.ReadAll(req.Body)
+	if err != nil {
+		panic(err)
+	}
+
+	// Create a MediaEngine object to configure the supported codec
+	media := &webrtc.MediaEngine{}
+
+	// Set up the codecs you want to use.
+	if err = media.RegisterCodec(webrtc.RTPCodecParameters{
+		RTPCodecCapability: webrtc.RTPCodecCapability{
+			MimeType:     webrtc.MimeTypeH264,
+			ClockRate:    90000,
+			Channels:     0,
+			SDPFmtpLine:  "",
+			RTCPFeedback: nil,
+		},
+		PayloadType: 96,
+	}, webrtc.RTPCodecTypeVideo); err != nil {
+		panic(err)
+	}
+	if err = media.RegisterCodec(webrtc.RTPCodecParameters{
+		RTPCodecCapability: webrtc.RTPCodecCapability{
+			MimeType:     webrtc.MimeTypeOpus,
+			ClockRate:    48000,
+			Channels:     2,
+			SDPFmtpLine:  "",
+			RTCPFeedback: nil,
+		},
+		PayloadType: 97,
+	}, webrtc.RTPCodecTypeAudio); err != nil {
+		panic(err)
+	}
+
+	// Create an InterceptorRegistry. This is the user configurable RTP/RTCP Pipeline.
+	ir := &interceptor.Registry{}
+
+	// Use the default set of Interceptors
+	if err = webrtc.RegisterDefaultInterceptors(media, ir); err != nil {
+		panic(err)
+	}
+
+	// We want TWCC in case the subscriber supports it
+	if err = webrtc.ConfigureTWCCHeaderExtensionSender(media, ir); err != nil {
+		panic(err)
+	}
+
+	// Create the API object with the MediaEngine
+	api := webrtc.NewAPI(webrtc.WithMediaEngine(media), webrtc.WithInterceptorRegistry(ir))
+
+	// Create a new RTCPeerConnection
+	peerConnection, err := api.NewPeerConnection(peerConnectionConfiguration)
+	if err != nil {
+		panic(err)
+	}
+
+	// Add Video Track that is being written to from WHIP Session
+	rtpSenderVideo, err := peerConnection.AddTrack(videoTrack)
+	if err != nil {
+		panic(err)
+	}
+	// Add Audio Track that is being written to from WHIP Session
+	rtpSenderAudio, err := peerConnection.AddTrack(audioTrack)
+	if err != nil {
+		panic(err)
+	}
+
+	// Read incoming RTCP packets for video
+	// 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 := rtpSenderVideo.Read(rtcpBuf); rtcpErr != nil {
+				return
+			}
+		}
+	}()
+
+	// Read incoming RTCP packets for audio
+	go func() {
+		rtcpBuf := make([]byte, 1500)
+		for {
+			if _, _, rtcpErr := rtpSenderAudio.Read(rtcpBuf); rtcpErr != nil {
+				return
+			}
+		}
+	}()
+
+	// Send answer via HTTP Response
+	writeAnswer(res, peerConnection, offer, "/whep")
+}
+
+func writeAnswer(res http.ResponseWriter, peerConnection *webrtc.PeerConnection, offer []byte, path string) {
+	// 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("ICE Connection State has changed: %s\n", connectionState.String())
+
+		if connectionState == webrtc.ICEConnectionStateFailed {
+			_ = peerConnection.Close()
+		}
+	})
+
+	if err := peerConnection.SetRemoteDescription(webrtc.SessionDescription{
+		Type: webrtc.SDPTypeOffer, SDP: string(offer),
+	}); err != nil {
+		panic(err)
+	}
+
+	// Create channel that is blocked until ICE Gathering is complete
+	gatherComplete := webrtc.GatheringCompletePromise(peerConnection)
+
+	// Create answer
+	answer, err := peerConnection.CreateAnswer(nil)
+	if err != nil {
+		panic(err)
+	} else 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
+
+	// WHIP+WHEP expects a Location header and a HTTP Status Code of 201
+	res.Header().Add("Location", path)
+	res.WriteHeader(http.StatusCreated)
+
+	// Write Answer with Candidates as HTTP Response
+	fmt.Fprint(res, peerConnection.LocalDescription().SDP) //nolint: errcheck
+}