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package webrtc
import (
"encoding/json"
"fmt"
"github.com/pion/interceptor"
"io/ioutil"
"net/http"
"regexp"
"strings"
"sync"
"time"
"github.com/Monibuca/engine/v3"
"github.com/Monibuca/plugin-webrtc/v3/webrtc"
"github.com/Monibuca/utils/v3"
"github.com/pion/rtcp"
. "github.com/pion/webrtc/v3"
"github.com/pion/webrtc/v3/pkg/media"
)
var config = struct {
ICEServers []string
PublicIP []string
PortMin uint16
PortMax uint16
PLI time.Duration
}{nil, nil, 0, 0, 2000}
// }{[]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
reg_level = regexp.MustCompile("profile-level-id=(4.+f)")
api *API
)
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() {
engine.InstallPlugin(&engine.PluginConfig{
Config: &config,
Name: "WebRTC",
Run: run,
})
}
type WebRTC struct {
*PeerConnection
}
func (rtc *WebRTC) Publish(streamPath string) bool {
if _, err := rtc.AddTransceiverFromKind(RTPCodecTypeVideo); err != nil {
if err != nil {
utils.Println("AddTransceiverFromKind video", err)
return false
}
}
if _, err := rtc.AddTransceiverFromKind(RTPCodecTypeAudio); err != nil {
if err != nil {
utils.Println("AddTransceiverFromKind audio", err)
return false
}
}
stream := &engine.Stream{
Type: "WebRTC",
StreamPath: streamPath,
}
if stream.Publish() {
rtc.OnICEConnectionStateChange(func(connectionState ICEConnectionState) {
utils.Printf("%s Connection State has changed %s ", streamPath, connectionState.String())
switch connectionState {
case ICEConnectionStateDisconnected, ICEConnectionStateFailed:
stream.Close()
}
})
//f, _ := os.OpenFile("resource/live/rtc.h264", os.O_TRUNC|os.O_WRONLY, 0666)
rtc.OnTrack(func(track *TrackRemote, receiver *RTPReceiver) {
if codec := track.Codec(); track.Kind() == RTPCodecTypeAudio {
var at *engine.RTPAudio
switch codec.PayloadType {
case 8:
at = stream.NewRTPAudio(7)
at.SoundRate = 8000
at.SoundSize = 16
at.Channels = 1
at.ExtraData = []byte{(at.CodecID << 4) | (1 << 1)}
case 0:
at = stream.NewRTPAudio(8)
at.SoundRate = 8000
at.SoundSize = 16
at.Channels = 1
at.ExtraData = []byte{(at.CodecID << 4) | (1 << 1)}
default:
return
}
for {
b := make([]byte, 1460)
if i, _, err := track.Read(b); err == nil {
at.Push(b[:i])
} else {
return
}
}
} else {
go func() {
ticker := time.NewTicker(time.Millisecond * config.PLI)
for {
select {
case <-ticker.C:
if rtcpErr := rtc.WriteRTCP([]rtcp.Packet{&rtcp.PictureLossIndication{MediaSSRC: uint32(track.SSRC())}}); rtcpErr != nil {
fmt.Println(rtcpErr)
}
case <-stream.Done():
return
}
}
}()
vt := stream.NewRTPVideo(7)
for {
b := make([]byte, 1460)
if i, _, err := track.Read(b); err == nil {
vt.Push(b[:i])
} else {
return
}
}
}
})
} 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
}
gatherComplete := GatheringCompletePromise(rtc.PeerConnection)
if err := rtc.SetLocalDescription(answer); err != nil {
return nil, err
}
<-gatherComplete
if bytes, err := json.Marshal(rtc.LocalDescription()); err != nil {
utils.Println(err)
return bytes, err
} else {
return bytes, nil
}
}
func run() {
var m MediaEngine
var s SettingEngine
//m.RegisterDefaultCodecs()
webrtc.RegisterCodecs(&m)
i := &interceptor.Registry{}
if len(config.PublicIP) > 0 {
s.SetNAT1To1IPs(config.PublicIP, ICECandidateTypeHost)
}
if config.PortMin > 0 && config.PortMax > 0 {
s.SetEphemeralUDPPortRange(config.PortMin, config.PortMax)
}
if len(config.PublicIP) > 0 {
s.SetNAT1To1IPs(config.PublicIP, ICECandidateTypeHost)
}
s.SetNetworkTypes([]NetworkType{NetworkTypeUDP4, NetworkTypeUDP6})
if err := RegisterDefaultInterceptors(&m, i); err != nil {
panic(err)
}
api := NewAPI(WithMediaEngine(&m),
WithInterceptorRegistry(i), WithSettingEngine(s))
http.HandleFunc("/api/webrtc/play", func(w http.ResponseWriter, r *http.Request) {
utils.CORS(w, r)
w.Header().Set("Content-Type", "application/json")
streamPath := r.URL.Query().Get("streamPath")
var offer SessionDescription
var rtc WebRTC
sub := engine.Subscriber{
ID: r.RemoteAddr,
Type: "WebRTC",
}
bytes, err := ioutil.ReadAll(r.Body)
defer func() {
if err != nil {
utils.Println(err)
fmt.Fprintf(w, `{"errmsg":"%s"}`, err)
return
}
}()
if err != nil {
return
}
if err = json.Unmarshal(bytes, &offer); err != nil {
return
}
if err = sub.Subscribe(streamPath); err != nil {
return
}
if rtc.PeerConnection, err = api.NewPeerConnection(Configuration{}); err != nil {
return
}
rtc.OnICECandidate(func(ice *ICECandidate) {
if ice != nil {
utils.Println(ice.ToJSON().Candidate)
}
})
if err = rtc.SetRemoteDescription(offer); err != nil {
return
}
vt := sub.WaitVideoTrack("h264")
at := sub.WaitAudioTrack("pcma", "pcmu")
var videoTrack *TrackLocalStaticSample
var rtpSender *RTPSender
if vt != nil {
pli := "42001f"
pli = fmt.Sprintf("%x", vt.ExtraData.NALUs[0][1:4])
if !strings.Contains(offer.SDP, pli) {
pli = reg_level.FindAllStringSubmatch(offer.SDP, -1)[0][1]
}
if videoTrack, err = NewTrackLocalStaticSample(RTPCodecCapability{MimeType: MimeTypeH264, SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=" + pli}, "video", "m7s"); err != nil {
return
}
rtpSender, err = rtc.AddTrack(videoTrack)
if err != nil {
return
}
var lastTimeStampV uint32
sub.OnVideo = func(ts uint32, pack *engine.VideoPack) {
var s uint32 = 40
if lastTimeStampV > 0 {
s = ts - lastTimeStampV
}
lastTimeStampV = ts
if pack.IDR {
err = videoTrack.WriteSample(media.Sample{
Data: vt.ExtraData.NALUs[0],
})
err = videoTrack.WriteSample(media.Sample{
Data: vt.ExtraData.NALUs[1],
})
}
for _, nalu := range pack.NALUs {
err = videoTrack.WriteSample(media.Sample{
Data: nalu,
Duration: time.Millisecond * time.Duration(s),
})
}
}
go func() {
rtcpBuf := make([]byte, 1500)
for {
if n, _, rtcpErr := rtpSender.Read(rtcpBuf); rtcpErr != nil {
return
} else {
if p, err := rtcp.Unmarshal(rtcpBuf[:n]); err == nil {
for _, pp := range p {
switch pp.(type) {
case *rtcp.PictureLossIndication:
fmt.Println("PictureLossIndication")
}
}
}
}
}
}()
}
if at != nil {
var audioTrack *TrackLocalStaticSample
audioMimeType := MimeTypePCMA
if at.CodecID == 8 {
audioMimeType = MimeTypePCMU
}
if audioTrack, err = NewTrackLocalStaticSample(RTPCodecCapability{audioMimeType, 8000, 0, "", nil}, "audio", "m7s"); err != nil {
return
}
if _, err = rtc.AddTrack(audioTrack); err != nil {
return
}
var lastTimeStampA uint32
sub.OnAudio = func(ts uint32, pack *engine.AudioPack) {
var s uint32
if lastTimeStampA > 0 {
s = ts - lastTimeStampA
}
lastTimeStampA = ts
audioTrack.WriteSample(media.Sample{
Data: pack.Raw, Duration: time.Millisecond * time.Duration(s),
})
}
}
if bytes, err := rtc.GetAnswer(); err == nil {
w.Write(bytes)
rtc.OnConnectionStateChange(func(pcs PeerConnectionState) {
utils.Printf("%s Connection State has changed %s ", streamPath, pcs.String())
switch pcs {
case PeerConnectionStateConnected:
if at != nil {
go sub.PlayAudio(at)
}
if vt != nil {
go sub.PlayVideo(vt)
}
case PeerConnectionStateDisconnected, PeerConnectionStateFailed:
sub.Close()
rtc.PeerConnection.Close()
}
})
} else {
return
}
})
http.HandleFunc("/api/webrtc/publish", func(w http.ResponseWriter, r *http.Request) {
utils.CORS(w, r)
streamPath := r.URL.Query().Get("streamPath")
offer := SessionDescription{}
bytes, err := ioutil.ReadAll(r.Body)
err = json.Unmarshal(bytes, &offer)
if err != nil {
utils.Println(err)
return
}
rtc := new(WebRTC)
if rtc.PeerConnection, err = api.NewPeerConnection(Configuration{}); err != nil {
return
}
if rtc.Publish(streamPath) {
if err := rtc.SetRemoteDescription(offer); err != nil {
utils.Println(err)
return
}
if bytes, err = rtc.GetAnswer(); err == nil {
w.Write(bytes)
} else {
utils.Println("GetAnswer:", err)
w.Write([]byte(err.Error()))
return
}
} else {
w.Write([]byte("bad name"))
}
})
}