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164 lines
3.7 KiB
164 lines
3.7 KiB
package webrtc
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import (
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"encoding/json"
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"fmt"
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"io/ioutil"
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"net/http"
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"time"
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. "github.com/Monibuca/engine/v2"
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rtsp "github.com/Monibuca/plugin-rtsp"
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"github.com/pion/rtcp"
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. "github.com/pion/webrtc/v2"
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)
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var config = &struct {
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ICEServers []string
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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 m = MediaEngine{}
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var api *API
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func init() {
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m.RegisterCodec(NewRTPH264Codec(DefaultPayloadTypeH264, 90000))
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//m.RegisterCodec(NewRTPPCMUCodec(DefaultPayloadTypePCMU, 8000))
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api = NewAPI(WithMediaEngine(m))
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InstallPlugin(&PluginConfig{
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Config: config,
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Name: "WebRTC",
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Type: PLUGIN_PUBLISHER | PLUGIN_SUBSCRIBER,
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Run: run,
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})
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}
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type WebRTC struct {
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rtsp.RTSP
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*PeerConnection
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}
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func (rtc *WebRTC) Publish(streamPath string) bool {
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peerConnection, err := api.NewPeerConnection(Configuration{
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ICEServers: []ICEServer{
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{
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URLs: config.ICEServers,
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},
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},
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})
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if _, err = peerConnection.AddTransceiverFromKind(RTPCodecTypeVideo); err != nil {
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if err != nil {
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Println(err)
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return false
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}
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}
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if err != nil {
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return false
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}
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peerConnection.OnICEConnectionStateChange(func(connectionState ICEConnectionState) {
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fmt.Printf("Connection State has changed %s \n", connectionState.String())
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if connectionState == ICEConnectionStateConnected {
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fmt.Println("Ctrl+C the remote client to stop the demo")
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} else if connectionState == ICEConnectionStateFailed ||
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connectionState == ICEConnectionStateDisconnected {
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fmt.Println("Done forwarding")
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//TODO
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}
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})
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rtc.PeerConnection = peerConnection
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if rtc.RTSP.Publish(streamPath) {
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rtc.Stream.Type = "WebRTC"
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peerConnection.OnTrack(func(track *Track, receiver *RTPReceiver) {
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defer rtc.Close()
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go func() {
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ticker := time.NewTicker(time.Second * 2)
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select {
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case <-ticker.C:
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if rtcpErr := peerConnection.WriteRTCP([]rtcp.Packet{&rtcp.PictureLossIndication{MediaSSRC: track.SSRC()}}); rtcpErr != nil {
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fmt.Println(rtcpErr)
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}
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case <-rtc.Done():
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return
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}
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}()
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pack := &rtsp.RTPPack{
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Type: rtsp.RTPType(track.Kind() - 1),
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}
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for b := make([]byte, 1460); ; rtc.HandleRTP(pack) {
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i, err := track.Read(b)
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if err != nil {
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return
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}
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if err = pack.Unmarshal(b[:i]); err != nil {
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return
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}
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}
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})
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} else {
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return false
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}
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return true
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}
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func run() {
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http.HandleFunc("/webrtc/answer", func(w http.ResponseWriter, r *http.Request) {
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streamPath := r.URL.Query().Get("streamPath")
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offer := SessionDescription{}
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bytes, err := ioutil.ReadAll(r.Body)
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err = json.Unmarshal(bytes, &offer)
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if err != nil {
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Println(err)
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return
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}
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rtc := new(WebRTC)
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if rtc.Publish(streamPath) {
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// Set the remote SessionDescription
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if err = rtc.SetRemoteDescription(offer); err != nil {
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Println(err)
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return
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}
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// Create answer
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answer, err := rtc.CreateAnswer(nil)
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if err != nil {
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Println(err)
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return
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}
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// Sets the LocalDescription, and starts our UDP listeners
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if err = rtc.SetLocalDescription(answer); err != nil {
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Println(err)
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return
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}
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if bytes, err = json.Marshal(answer); err != nil {
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Println(err)
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return
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}
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w.Write(bytes)
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}
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})
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}
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