package webrtc import ( "encoding/json" "fmt" "io/ioutil" "net/http" "sync" "time" . "github.com/Monibuca/engine/v2" "github.com/Monibuca/engine/v2/avformat" . "github.com/Monibuca/plugin-rtp" "github.com/pion/rtcp" . "github.com/pion/webrtc/v2" "github.com/pion/webrtc/v2/pkg/media" ) var config struct { ICEServers []string } // }{[]string{ // "stun:stun.ekiga.net", // "stun:stun.ideasip.com", // "stun:stun.schlund.de", // "stun:stun.stunprotocol.org:3478", // "stun:stun.voiparound.com", // "stun:stun.voipbuster.com", // "stun:stun.voipstunt.com", // "stun:stun.voxgratia.org", // "stun:stun.services.mozilla.com", // "stun:stun.xten.com", // "stun:stun.softjoys.com", // "stun:stunserver.org", // "stun:stun.schlund.de", // "stun:stun.rixtelecom.se", // "stun:stun.iptel.org", // "stun:stun.ideasip.com", // "stun:stun.fwdnet.net", // "stun:stun.ekiga.net", // "stun:stun01.sipphone.com", // }} // type udpConn struct { // conn *net.UDPConn // port int // } var playWaitList WaitList type WaitList struct { m map[string]*WebRTC l sync.Mutex } func (wl *WaitList) Set(k string, v *WebRTC) { wl.l.Lock() defer wl.l.Unlock() if wl.m == nil { wl.m = make(map[string]*WebRTC) } wl.m[k] = v } func (wl *WaitList) Get(k string) *WebRTC { wl.l.Lock() defer wl.l.Unlock() defer delete(wl.m, k) return wl.m[k] } func init() { InstallPlugin(&PluginConfig{ Config: &config, Name: "WebRTC", Type: PLUGIN_PUBLISHER | PLUGIN_SUBSCRIBER, Run: run, }) } type WebRTC struct { RTP *PeerConnection RemoteAddr string videoTrack *Track m MediaEngine api *API payloader avformat.H264 // codecs.H264Packet // *os.File } func (rtc *WebRTC) Play(streamPath string) bool { var sub Subscriber sub.ID = rtc.RemoteAddr sub.Type = "WebRTC" var lastTimeStamp uint32 sub.OnData = func(packet *avformat.SendPacket) error { if packet.Type == avformat.FLV_TAG_TYPE_AUDIO { return nil } if packet.IsSequence { rtc.payloader.PPS = sub.PPS rtc.payloader.SPS = sub.SPS } else { var s uint32 if lastTimeStamp > 0 { s = packet.Timestamp - lastTimeStamp } lastTimeStamp = packet.Timestamp rtc.videoTrack.WriteSample(media.Sample{ Data: packet.Payload, Samples: s * 90, }) // if packet.IsKeyFrame { // rtc.videoTrack.WriteSample(media.Sample{ // Data: sub.SPS, // Samples: 0, // }) // rtc.videoTrack.WriteSample(media.Sample{ // Data: sub.PPS, // Samples: 0, // }) // } // for payload := packet.Payload[5:]; len(payload) > 4; { // var naulLen = int(util.BigEndian.Uint32(payload)) // payload = payload[4:] // rtc.videoTrack.WriteSample(media.Sample{ // Data: payload[:naulLen], // Samples: s * 90, // }) // s = 0 // payload = payload[naulLen:] // } } return nil } // go sub.Subscribe(streamPath) rtc.OnICEConnectionStateChange(func(connectionState ICEConnectionState) { Printf("%s Connection State has changed %s ", streamPath, connectionState.String()) switch connectionState { case ICEConnectionStateDisconnected: sub.Close() case ICEConnectionStateConnected: //rtc.videoTrack = rtc.GetSenders()[0].Track() sub.Subscribe(streamPath) } }) return true } func (rtc *WebRTC) Publish(streamPath string) bool { rtc.m.RegisterCodec(NewRTPCodec(RTPCodecTypeVideo, H264, 90000, 0, "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f", DefaultPayloadTypeH264, new(avformat.H264))) //m.RegisterCodec(NewRTPPCMUCodec(DefaultPayloadTypePCMU, 8000)) rtc.api = NewAPI(WithMediaEngine(rtc.m)) peerConnection, err := rtc.api.NewPeerConnection(Configuration{ ICEServers: []ICEServer{ { URLs: config.ICEServers, }, }, }) if _, err = peerConnection.AddTransceiverFromKind(RTPCodecTypeVideo); err != nil { if err != nil { Println(err) return false } } if err != nil { return false } peerConnection.OnICEConnectionStateChange(func(connectionState ICEConnectionState) { Printf("%s Connection State has changed %s ", streamPath, connectionState.String()) switch connectionState { case ICEConnectionStateDisconnected, ICEConnectionStateFailed: if rtc.Stream != nil { rtc.Stream.Close() } } }) rtc.PeerConnection = peerConnection if rtc.RTP.Publish(streamPath) { //f, _ := os.OpenFile("resource/live/rtc.h264", os.O_TRUNC|os.O_WRONLY, 0666) rtc.Stream.Type = "WebRTC" peerConnection.OnTrack(func(track *Track, receiver *RTPReceiver) { defer rtc.Stream.Close() go func() { ticker := time.NewTicker(time.Second * 2) select { case <-ticker.C: if rtcpErr := peerConnection.WriteRTCP([]rtcp.Packet{&rtcp.PictureLossIndication{MediaSSRC: track.SSRC()}}); rtcpErr != nil { fmt.Println(rtcpErr) } case <-rtc.Done(): return } }() pack := &RTPPack{ Type: RTPType(track.Kind() - 1), } for b := make([]byte, 1460); ; rtc.PushPack(pack) { i, err := track.Read(b) if err != nil { return } if err = pack.Unmarshal(b[:i]); err != nil { return } // rtc.Unmarshal(pack.Payload) // f.Write(bytes) } }) } else { return false } return true } func (rtc *WebRTC) GetAnswer() ([]byte, error) { // Sets the LocalDescription, and starts our UDP listeners answer, err := rtc.CreateAnswer(nil) if err != nil { return nil, err } if err := rtc.SetLocalDescription(answer); err != nil { Println(err) return nil, err } if bytes, err := json.Marshal(answer); err != nil { Println(err) return bytes, err } else { return bytes, nil } } func run() { http.HandleFunc("/webrtc/play", func(w http.ResponseWriter, r *http.Request) { streamPath := r.URL.Query().Get("streamPath") var offer SessionDescription var rtc WebRTC bytes, err := ioutil.ReadAll(r.Body) defer func() { if err != nil { Println(err) fmt.Fprintf(w, `{"errmsg":"%s"}`, err) return } rtc.Play(streamPath) }() if err != nil { return } if err = json.Unmarshal(bytes, &offer); err != nil { return } // pli := "42001f" // if stream := FindStream(streamPath); stream != nil { // pli = fmt.Sprintf("%x", stream.SPS[1:4]) // } rtc.m.RegisterCodec(NewRTPCodec(RTPCodecTypeVideo, H264, 90000, 0, "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f", DefaultPayloadTypeH264, &rtc.payloader)) //m.RegisterCodec(NewRTPPCMUCodec(DefaultPayloadTypePCMU, 8000)) rtc.api = NewAPI(WithMediaEngine(rtc.m)) peerConnection, err := rtc.api.NewPeerConnection(Configuration{ // ICEServers: []ICEServer{ // { // URLs: config.ICEServers, // }, // }, }) rtc.PeerConnection = peerConnection rtc.OnICECandidate(func(ice *ICECandidate) { if ice != nil { Println(ice.ToJSON().Candidate) } }) // if r, err := peerConnection.AddTransceiverFromKind(RTPCodecTypeVideo); err == nil { // rtc.videoTrack = r.Sender().Track() // } else { // Println(err) // } if err != nil { return } rtc.RemoteAddr = r.RemoteAddr if err = rtc.SetRemoteDescription(offer); err != nil { return } rtc.m.PopulateFromSDP(offer) // var vpayloadType uint8 = 0 // for _, videoCodec := range rtc.m.GetCodecsByKind(RTPCodecTypeVideo) { // if videoCodec.Name == H264 { // vpayloadType = videoCodec.PayloadType // videoCodec.Payloader = &rtc.payloader // Printf("H264 fmtp %v", videoCodec.SDPFmtpLine) // break // } // } if rtc.videoTrack, err = rtc.NewTrack(DefaultPayloadTypeH264, 8, "video", "monibuca"); err != nil { return } if _, err = rtc.AddTrack(rtc.videoTrack); err != nil { return } if bytes, err := rtc.GetAnswer(); err == nil { w.Write(bytes) } else { return } }) http.HandleFunc("/webrtc/publish", func(w http.ResponseWriter, r *http.Request) { streamPath := r.URL.Query().Get("streamPath") offer := SessionDescription{} bytes, err := ioutil.ReadAll(r.Body) err = json.Unmarshal(bytes, &offer) if err != nil { Println(err) return } rtc := new(WebRTC) rtc.RemoteAddr = r.RemoteAddr if rtc.Publish(streamPath) { if err := rtc.SetRemoteDescription(offer); err != nil { Println(err) return } if bytes, err = rtc.GetAnswer(); err == nil { w.Write(bytes) } else { Println(err) w.Write([]byte(err.Error())) return } } else { w.Write([]byte("bad name")) } }) }