github.com/pion/webrtc/v4@v4.0.1/examples/rtp-to-webrtc/main.go (about)

     1  // SPDX-FileCopyrightText: 2023 The Pion community <https://pion.ly>
     2  // SPDX-License-Identifier: MIT
     3  
     4  //go:build !js
     5  // +build !js
     6  
     7  // rtp-to-webrtc demonstrates how to consume a RTP stream video UDP, and then send to a WebRTC client.
     8  package main
     9  
    10  import (
    11  	"bufio"
    12  	"encoding/base64"
    13  	"encoding/json"
    14  	"errors"
    15  	"fmt"
    16  	"io"
    17  	"net"
    18  	"os"
    19  	"strings"
    20  
    21  	"github.com/pion/webrtc/v4"
    22  )
    23  
    24  func main() {
    25  	peerConnection, err := webrtc.NewPeerConnection(webrtc.Configuration{
    26  		ICEServers: []webrtc.ICEServer{
    27  			{
    28  				URLs: []string{"stun:stun.l.google.com:19302"},
    29  			},
    30  		},
    31  	})
    32  	if err != nil {
    33  		panic(err)
    34  	}
    35  
    36  	// Open a UDP Listener for RTP Packets on port 5004
    37  	listener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: 5004})
    38  	if err != nil {
    39  		panic(err)
    40  	}
    41  
    42  	// Increase the UDP receive buffer size
    43  	// Default UDP buffer sizes vary on different operating systems
    44  	bufferSize := 300000 // 300KB
    45  	err = listener.SetReadBuffer(bufferSize)
    46  	if err != nil {
    47  		panic(err)
    48  	}
    49  
    50  	defer func() {
    51  		if err = listener.Close(); err != nil {
    52  			panic(err)
    53  		}
    54  	}()
    55  
    56  	// Create a video track
    57  	videoTrack, err := webrtc.NewTrackLocalStaticRTP(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeVP8}, "video", "pion")
    58  	if err != nil {
    59  		panic(err)
    60  	}
    61  	rtpSender, err := peerConnection.AddTrack(videoTrack)
    62  	if err != nil {
    63  		panic(err)
    64  	}
    65  
    66  	// Read incoming RTCP packets
    67  	// Before these packets are returned they are processed by interceptors. For things
    68  	// like NACK this needs to be called.
    69  	go func() {
    70  		rtcpBuf := make([]byte, 1500)
    71  		for {
    72  			if _, _, rtcpErr := rtpSender.Read(rtcpBuf); rtcpErr != nil {
    73  				return
    74  			}
    75  		}
    76  	}()
    77  
    78  	// Set the handler for ICE connection state
    79  	// This will notify you when the peer has connected/disconnected
    80  	peerConnection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
    81  		fmt.Printf("Connection State has changed %s \n", connectionState.String())
    82  
    83  		if connectionState == webrtc.ICEConnectionStateFailed {
    84  			if closeErr := peerConnection.Close(); closeErr != nil {
    85  				panic(closeErr)
    86  			}
    87  		}
    88  	})
    89  
    90  	// Wait for the offer to be pasted
    91  	offer := webrtc.SessionDescription{}
    92  	decode(readUntilNewline(), &offer)
    93  
    94  	// Set the remote SessionDescription
    95  	if err = peerConnection.SetRemoteDescription(offer); err != nil {
    96  		panic(err)
    97  	}
    98  
    99  	// Create answer
   100  	answer, err := peerConnection.CreateAnswer(nil)
   101  	if err != nil {
   102  		panic(err)
   103  	}
   104  
   105  	// Create channel that is blocked until ICE Gathering is complete
   106  	gatherComplete := webrtc.GatheringCompletePromise(peerConnection)
   107  
   108  	// Sets the LocalDescription, and starts our UDP listeners
   109  	if err = peerConnection.SetLocalDescription(answer); err != nil {
   110  		panic(err)
   111  	}
   112  
   113  	// Block until ICE Gathering is complete, disabling trickle ICE
   114  	// we do this because we only can exchange one signaling message
   115  	// in a production application you should exchange ICE Candidates via OnICECandidate
   116  	<-gatherComplete
   117  
   118  	// Output the answer in base64 so we can paste it in browser
   119  	fmt.Println(encode(peerConnection.LocalDescription()))
   120  
   121  	// Read RTP packets forever and send them to the WebRTC Client
   122  	inboundRTPPacket := make([]byte, 1600) // UDP MTU
   123  	for {
   124  		n, _, err := listener.ReadFrom(inboundRTPPacket)
   125  		if err != nil {
   126  			panic(fmt.Sprintf("error during read: %s", err))
   127  		}
   128  
   129  		if _, err = videoTrack.Write(inboundRTPPacket[:n]); err != nil {
   130  			if errors.Is(err, io.ErrClosedPipe) {
   131  				// The peerConnection has been closed.
   132  				return
   133  			}
   134  
   135  			panic(err)
   136  		}
   137  	}
   138  }
   139  
   140  // Read from stdin until we get a newline
   141  func readUntilNewline() (in string) {
   142  	var err error
   143  
   144  	r := bufio.NewReader(os.Stdin)
   145  	for {
   146  		in, err = r.ReadString('\n')
   147  		if err != nil && !errors.Is(err, io.EOF) {
   148  			panic(err)
   149  		}
   150  
   151  		if in = strings.TrimSpace(in); len(in) > 0 {
   152  			break
   153  		}
   154  	}
   155  
   156  	fmt.Println("")
   157  	return
   158  }
   159  
   160  // JSON encode + base64 a SessionDescription
   161  func encode(obj *webrtc.SessionDescription) string {
   162  	b, err := json.Marshal(obj)
   163  	if err != nil {
   164  		panic(err)
   165  	}
   166  
   167  	return base64.StdEncoding.EncodeToString(b)
   168  }
   169  
   170  // Decode a base64 and unmarshal JSON into a SessionDescription
   171  func decode(in string, obj *webrtc.SessionDescription) {
   172  	b, err := base64.StdEncoding.DecodeString(in)
   173  	if err != nil {
   174  		panic(err)
   175  	}
   176  
   177  	if err = json.Unmarshal(b, obj); err != nil {
   178  		panic(err)
   179  	}
   180  }