webrtc/examples/rtp-to-webrtc/main.go

// +build !js

package main

import (
	"fmt"
	"net"

	"github.com/pion/rtp"
	"github.com/pion/rtp/codecs"
	"github.com/pion/webrtc/v3"
	"github.com/pion/webrtc/v3/examples/internal/signal"
	"github.com/pion/webrtc/v3/pkg/media/samplebuilder"
)

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)
	}
	defer func() {
		if err = listener.Close(); err != nil {
			panic(err)
		}
	}()

	fmt.Println("Waiting for RTP Packets, please run GStreamer or ffmpeg now")

	// Listen for a single RTP Packet, we need this to determine the SSRC
	inboundRTPPacket := make([]byte, 1500) // UDP MTU
	n, _, err := listener.ReadFromUDP(inboundRTPPacket)
	if err != nil {
		panic(err)
	}

	// Unmarshal the incoming packet
	packet := &rtp.Packet{}
	if err = packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
		panic(err)
	}

	// Create a video track
	videoTrack, err := webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: "video/vp8"}, "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 retuned 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())
	})

	// 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()))

	videoBuilder := samplebuilder.New(10, &codecs.VP8Packet{}, 90000)

	// Read RTP packets forever and send them to the WebRTC Client
	for {
		inboundRTPPacket = make([]byte, 1500) // UDP MTU
		packet = &rtp.Packet{}

		n, _, err := listener.ReadFrom(inboundRTPPacket)
		if err != nil {
			panic(fmt.Sprintf("error during read: %s", err))
		}

		if err = packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
			panic(err)
		}

		videoBuilder.Push(packet)
		for {
			sample := videoBuilder.Pop()
			if sample == nil {
				break
			}

			if writeErr := videoTrack.WriteSample(*sample); writeErr != nil {
				panic(writeErr)
			}
		}
	}
}