github.com/pion/webrtc/v3@v3.2.24/examples/rtp-forwarder/main.go

// SPDX-FileCopyrightText: 2023 The Pion community <https://pion.ly>
// SPDX-License-Identifier: MIT

//go:build !js
// +build !js

// rtp-forwarder shows how to forward your webcam/microphone via RTP using Pion WebRTC.
package main

import (
	"errors"
	"fmt"
	"net"
	"os"

	"github.com/pion/interceptor"
	"github.com/pion/interceptor/pkg/intervalpli"
	"github.com/pion/rtp"
	"github.com/pion/webrtc/v3"
	"github.com/pion/webrtc/v3/examples/internal/signal"
)

type udpConn struct {
	conn        *net.UDPConn
	port        int
	payloadType uint8
}

// nolint:gocognit
func main() {
	// Everything below is the Pion WebRTC API! Thanks for using it ❤️.

	// Create a MediaEngine object to configure the supported codec
	m := &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 := m.RegisterCodec(webrtc.RTPCodecParameters{
		RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeVP8, ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: nil},
	}, webrtc.RTPCodecTypeVideo); err != nil {
		panic(err)
	}
	if err := m.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.
	i := &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)
	}
	i.Add(intervalPliFactory)

	// Use the default set of Interceptors
	if err = webrtc.RegisterDefaultInterceptors(m, i); err != nil {
		panic(err)
	}

	// Create the API object with the MediaEngine
	api := webrtc.NewAPI(webrtc.WithMediaEngine(m), webrtc.WithInterceptorRegistry(i))

	// 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 _, 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.TrackRemote, receiver *webrtc.RTPReceiver) {
		// Retrieve udp connection
		c, ok := udpConns[track.Kind().String()]
		if !ok {
			return
		}

		b := make([]byte, 1500)
		rtpPacket := &rtp.Packet{}
		for {
			// Read
			n, _, readErr := track.Read(b)
			if readErr != nil {
				panic(readErr)
			}

			// Unmarshal the packet and update the PayloadType
			if err = rtpPacket.Unmarshal(b[:n]); err != nil {
				panic(err)
			}
			rtpPacket.PayloadType = c.payloadType

			// Marshal into original buffer with updated PayloadType
			if n, err = rtpPacket.MarshalTo(b); err != nil {
				panic(err)
			}

			// Write
			if _, writeErr := c.conn.Write(b[: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(s webrtc.PeerConnectionState) {
		fmt.Printf("Peer Connection State has changed: %s\n", s.String())

		if s == 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)
		}
	})

	// 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)
	}

	// 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(signal.Encode(*peerConnection.LocalDescription()))

	// Block forever
	select {}
}