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172 lines
4.7 KiB
172 lines
4.7 KiB
package webrtc
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import (
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"io/ioutil"
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"m7s.live/engine/v4"
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"net"
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"net/http"
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"regexp"
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"time"
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"github.com/pion/interceptor"
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. "github.com/pion/webrtc/v3"
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"m7s.live/engine/v4/config"
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"m7s.live/plugin/webrtc/v4/webrtc"
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)
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// }{[]string{
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// "stun:stun.ekiga.net",
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// "stun:stun.ideasip.com",
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// "stun:stun.schlund.de",
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// "stun:stun.stunprotocol.org:3478",
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// "stun:stun.voiparound.com",
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// "stun:stun.voipbuster.com",
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// "stun:stun.voipstunt.com",
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// "stun:stun.voxgratia.org",
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// "stun:stun.services.mozilla.com",
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// "stun:stun.xten.com",
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// "stun:stun.softjoys.com",
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// "stun:stunserver.org",
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// "stun:stun.schlund.de",
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// "stun:stun.rixtelecom.se",
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// "stun:stun.iptel.org",
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// "stun:stun.ideasip.com",
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// "stun:stun.fwdnet.net",
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// "stun:stun.ekiga.net",
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// "stun:stun01.sipphone.com",
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// }}
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// type udpConn struct {
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// conn *net.UDPConn
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// port int
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// }
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var (
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reg_level = regexp.MustCompile("profile-level-id=(4.+f)")
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)
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type WebRTCConfig struct {
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config.Publish
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config.Subscribe
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ICEServers []string
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PublicIP []string
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PortMin uint16
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PortMax uint16
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InvitePortFixed bool
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IceUdpMux int
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PLI time.Duration
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m MediaEngine
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s SettingEngine
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api *API
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}
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func (conf *WebRTCConfig) OnEvent(event any) {
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switch event.(type) {
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case engine.FirstConfig:
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webrtc.RegisterCodecs(&conf.m)
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i := &interceptor.Registry{}
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if len(conf.PublicIP) > 0 {
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conf.s.SetNAT1To1IPs(conf.PublicIP, ICECandidateTypeHost)
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}
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if conf.PortMin > 0 && conf.PortMax > 0 {
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conf.s.SetEphemeralUDPPortRange(conf.PortMin, conf.PortMax)
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}
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// 是否多路复用UDP端口
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if conf.InvitePortFixed {
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// 创建共享WEBRTC端口 默认9000
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udpListener, _ := net.ListenUDP("udp", &net.UDPAddr{
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IP: net.IP{0, 0, 0, 0},
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Port: conf.IceUdpMux,
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})
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conf.s.SetICEUDPMux(NewICEUDPMux(nil, udpListener))
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}
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conf.s.SetNetworkTypes([]NetworkType{NetworkTypeUDP4, NetworkTypeUDP6})
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if err := RegisterDefaultInterceptors(&conf.m, i); err != nil {
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panic(err)
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}
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conf.api = NewAPI(WithMediaEngine(&conf.m),
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WithInterceptorRegistry(i), WithSettingEngine(conf.s))
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}
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}
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func (conf *WebRTCConfig) Play_(w http.ResponseWriter, r *http.Request) {
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w.Header().Set("Content-Type", "application/sdp")
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streamPath := r.URL.Path[len("/webrtc/play/"):]
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bytes, err := ioutil.ReadAll(r.Body)
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var suber WebRTCSubscriber
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suber.SDP = string(bytes)
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if suber.PeerConnection, err = conf.api.NewPeerConnection(Configuration{}); err != nil {
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http.Error(w, err.Error(), http.StatusInternalServerError)
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return
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}
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suber.OnICECandidate(func(ice *ICECandidate) {
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if ice != nil {
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suber.Info(ice.ToJSON().Candidate)
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}
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})
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if err = suber.SetRemoteDescription(SessionDescription{Type: SDPTypeOffer, SDP: suber.SDP}); err != nil {
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http.Error(w, err.Error(), http.StatusInternalServerError)
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return
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}
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if err = plugin.Subscribe(streamPath, &suber); err != nil {
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http.Error(w, err.Error(), http.StatusBadRequest)
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return
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}
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if sdp, err := suber.GetAnswer(); err == nil {
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w.Write([]byte(sdp))
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} else {
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http.Error(w, err.Error(), http.StatusBadRequest)
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}
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}
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func (conf *WebRTCConfig) Push_(w http.ResponseWriter, r *http.Request) {
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streamPath := r.URL.Path[len("/webrtc/push/"):]
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w.Header().Set("Content-Type", "application/sdp")
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bytes, err := ioutil.ReadAll(r.Body)
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var puber WebRTCPublisher
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puber.SDP = string(bytes)
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if puber.PeerConnection, err = conf.api.NewPeerConnection(Configuration{}); err != nil {
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http.Error(w, err.Error(), http.StatusInternalServerError)
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return
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}
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puber.OnICECandidate(func(ice *ICECandidate) {
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if ice != nil {
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puber.Info(ice.ToJSON().Candidate)
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}
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})
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if _, err = puber.AddTransceiverFromKind(RTPCodecTypeVideo); err != nil {
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http.Error(w, err.Error(), http.StatusInternalServerError)
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return
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}
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if _, err = puber.AddTransceiverFromKind(RTPCodecTypeAudio); err != nil {
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http.Error(w, err.Error(), http.StatusInternalServerError)
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return
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}
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if err = plugin.Publish(streamPath, &puber); err != nil {
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http.Error(w, err.Error(), http.StatusBadRequest)
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return
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}
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if err := puber.SetRemoteDescription(SessionDescription{Type: SDPTypeOffer, SDP: puber.SDP}); err != nil {
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http.Error(w, err.Error(), http.StatusBadRequest)
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return
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}
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if answer, err := puber.GetAnswer(); err == nil {
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w.Write([]byte(answer))
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} else {
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http.Error(w, err.Error(), http.StatusBadRequest)
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return
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}
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}
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var webrtcConfig = &WebRTCConfig{
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InvitePortFixed: true, // 设备将流发送的端口,是否固定 on 发送流到多路复用端口 如9000 off 自动从 mix_port - max_port 之间的值中 选一个可以用的端口
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IceUdpMux: 9000, // 接收设备端rtp流的多路复用端口
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PLI: time.Second * 2,
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}
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var plugin = engine.InstallPlugin(webrtcConfig)
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