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@ -1,41 +1,49 @@ |
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package webrtc |
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package webrtc |
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import ( |
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import ( |
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"bytes" |
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"encoding/json" |
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"encoding/json" |
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"fmt" |
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"fmt" |
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"io/ioutil" |
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"io/ioutil" |
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"net/http" |
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"net/http" |
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"sync" |
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"time" |
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"time" |
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. "github.com/Monibuca/engine/v2" |
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. "github.com/Monibuca/engine/v2" |
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"github.com/Monibuca/engine/v2/avformat" |
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"github.com/Monibuca/engine/v2/util" |
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. "github.com/Monibuca/plugin-rtp" |
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. "github.com/Monibuca/plugin-rtp" |
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"github.com/pion/rtcp" |
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"github.com/pion/rtcp" |
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"github.com/pion/rtp" |
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"github.com/pion/rtp/codecs" |
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. "github.com/pion/webrtc/v2" |
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. "github.com/pion/webrtc/v2" |
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) |
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) |
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var config = &struct { |
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var config struct { |
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ICEServers []string |
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ICEServers []string |
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}{[]string{ |
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} |
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"stun:stun.ekiga.net", |
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"stun:stun.ideasip.com", |
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// }{[]string{
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"stun:stun.schlund.de", |
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// "stun:stun.ekiga.net",
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"stun:stun.stunprotocol.org:3478", |
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// "stun:stun.ideasip.com",
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"stun:stun.voiparound.com", |
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// "stun:stun.schlund.de",
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"stun:stun.voipbuster.com", |
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// "stun:stun.stunprotocol.org:3478",
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"stun:stun.voipstunt.com", |
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// "stun:stun.voiparound.com",
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"stun:stun.voxgratia.org", |
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// "stun:stun.voipbuster.com",
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"stun:stun.services.mozilla.com", |
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// "stun:stun.voipstunt.com",
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"stun:stun.xten.com", |
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// "stun:stun.voxgratia.org",
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"stun:stun.softjoys.com", |
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// "stun:stun.services.mozilla.com",
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"stun:stunserver.org", |
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// "stun:stun.xten.com",
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"stun:stun.schlund.de", |
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// "stun:stun.softjoys.com",
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"stun:stun.rixtelecom.se", |
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// "stun:stunserver.org",
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"stun:stun.iptel.org", |
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// "stun:stun.schlund.de",
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"stun:stun.ideasip.com", |
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// "stun:stun.rixtelecom.se",
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"stun:stun.fwdnet.net", |
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// "stun:stun.iptel.org",
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"stun:stun.ekiga.net", |
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// "stun:stun.ideasip.com",
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"stun:stun01.sipphone.com", |
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// "stun:stun.fwdnet.net",
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}} |
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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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// type udpConn struct {
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// conn *net.UDPConn
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// conn *net.UDPConn
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@ -44,13 +52,16 @@ var config = &struct { |
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var m MediaEngine |
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var m MediaEngine |
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var api *API |
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var api *API |
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var SSRC uint32 |
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var SSRCMap = make(map[string]uint32) |
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var ssrcLock sync.Mutex |
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func init() { |
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func init() { |
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m.RegisterCodec(NewRTPH264Codec(DefaultPayloadTypeH264, 90000)) |
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m.RegisterCodec(NewRTPH264Codec(DefaultPayloadTypeH264, 90000)) |
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//m.RegisterCodec(NewRTPPCMUCodec(DefaultPayloadTypePCMU, 8000))
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//m.RegisterCodec(NewRTPPCMUCodec(DefaultPayloadTypePCMU, 8000))
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api = NewAPI(WithMediaEngine(m)) |
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api = NewAPI(WithMediaEngine(m)) |
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InstallPlugin(&PluginConfig{ |
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InstallPlugin(&PluginConfig{ |
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Config: config, |
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Config: &config, |
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Name: "WebRTC", |
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Name: "WebRTC", |
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Type: PLUGIN_PUBLISHER | PLUGIN_SUBSCRIBER, |
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Type: PLUGIN_PUBLISHER | PLUGIN_SUBSCRIBER, |
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Run: run, |
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Run: run, |
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@ -60,9 +71,10 @@ func init() { |
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type WebRTC struct { |
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type WebRTC struct { |
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RTP |
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RTP |
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*PeerConnection |
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*PeerConnection |
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RemoteAddr string |
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} |
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} |
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func (rtc *WebRTC) Publish(streamPath string) bool { |
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func (rtc *WebRTC) Play(streamPath string) bool { |
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peerConnection, err := api.NewPeerConnection(Configuration{ |
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peerConnection, err := api.NewPeerConnection(Configuration{ |
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ICEServers: []ICEServer{ |
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ICEServers: []ICEServer{ |
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{ |
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{ |
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@ -79,15 +91,170 @@ func (rtc *WebRTC) Publish(streamPath string) bool { |
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if err != nil { |
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if err != nil { |
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return false |
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return false |
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} |
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} |
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rtc.PeerConnection = peerConnection |
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// Create a video track, using the same SSRC as the incoming RTP Packet
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ssrcLock.Lock() |
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ssrc, ok := SSRCMap[streamPath] |
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if !ok { |
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SSRC++ |
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ssrc = SSRC |
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SSRCMap[streamPath] = SSRC |
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} |
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ssrcLock.Unlock() |
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videoTrack, err := peerConnection.NewTrack(DefaultPayloadTypeH264, ssrc, "video", "monibuca") |
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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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if _, err = peerConnection.AddTrack(videoTrack); err != nil { |
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Println(err) |
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return false |
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} |
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peerConnection.OnICEConnectionStateChange(func(connectionState ICEConnectionState) { |
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peerConnection.OnICEConnectionStateChange(func(connectionState ICEConnectionState) { |
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Printf("%s Connection State has changed %s ", streamPath, connectionState.String()) |
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Printf("%s Connection State has changed %s ", streamPath, connectionState.String()) |
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switch connectionState { |
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switch connectionState { |
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case ICEConnectionStateDisconnected: |
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case ICEConnectionStateDisconnected: |
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rtc.Stream.Close() |
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if rtc.Stream != nil { |
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rtc.Stream.Close() |
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} |
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case ICEConnectionStateConnected: |
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var sequence uint16 |
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var sub Subscriber |
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sub.ID = rtc.RemoteAddr |
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sub.Type = "WebRTC" |
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nextHeader := func(ts uint32, marker bool) rtp.Header { |
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sequence++ |
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return rtp.Header{ |
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Version: 2, |
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SSRC: ssrc, |
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PayloadType: DefaultPayloadTypeH264, |
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SequenceNumber: sequence, |
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Timestamp: ts, |
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Marker: marker, |
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} |
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} |
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stapA := func(naul ...[]byte) []byte { |
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var buffer bytes.Buffer |
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buffer.WriteByte(24) |
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for _, n := range naul { |
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l := len(n) |
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buffer.WriteByte(byte(l >> 8)) |
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buffer.WriteByte(byte(l)) |
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buffer.Write(n) |
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} |
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return buffer.Bytes() |
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} |
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// aud := []byte{0x09, 0x30}
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sub.OnData = func(packet *avformat.SendPacket) error { |
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if packet.Type == avformat.FLV_TAG_TYPE_AUDIO { |
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return nil |
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} |
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if packet.IsSequence { |
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payload := packet.Payload[11:] |
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spsLen := int(payload[0])<<8 + int(payload[1]) |
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sps := payload[2:spsLen] |
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payload = payload[3+spsLen:] |
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ppsLen := int(payload[0])<<8 + int(payload[1]) |
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pps := payload[2:ppsLen] |
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if err := videoTrack.WriteRTP(&rtp.Packet{ |
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Header: nextHeader(0, false), |
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Payload: stapA(sps, pps), |
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}); err != nil { |
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return err |
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} |
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} else { |
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payload := packet.Payload[5:] |
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for { |
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var naulLen = int(util.BigEndian.Uint32(payload)) |
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payload = payload[4:] |
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_payload := payload[:naulLen] |
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if naulLen > 1000 { |
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part := _payload[:1000] |
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indicator := ((part[0] >> 5) << 5) | 28 |
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nalutype := part[0] & 31 |
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header := 128 | nalutype |
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part = part[1:] |
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marker := false |
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for { |
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if err := videoTrack.WriteRTP(&rtp.Packet{ |
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Header: nextHeader(packet.Timestamp*90, marker), |
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Payload: append([]byte{indicator, header}, part...), |
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}); err != nil { |
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return err |
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} |
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marker = true |
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if _payload == nil { |
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break |
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} |
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if len(_payload[1000:]) <= 1000 { |
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header = 64 | nalutype |
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part = _payload[1000:] |
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_payload = nil |
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} else { |
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header = nalutype |
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part = _payload[1000:] |
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_payload = part |
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} |
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} |
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} else { |
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if err := videoTrack.WriteRTP(&rtp.Packet{ |
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Header: nextHeader(packet.Timestamp*90, false), |
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Payload: _payload, |
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}); err != nil { |
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return err |
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} |
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} |
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if len(payload) < naulLen+4 { |
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break |
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} |
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payload = payload[naulLen:] |
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} |
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// if err := videoTrack.WriteRTP(&rtp.Packet{
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// Header: nextHeader(packet.Timestamp * 90),
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// Payload: aud,
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// }); err != nil {
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// return err
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// }
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} |
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return nil |
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} |
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sub.Subscribe(streamPath) |
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} |
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}) |
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return true |
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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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Printf("%s Connection State has changed %s ", streamPath, connectionState.String()) |
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switch connectionState { |
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case ICEConnectionStateDisconnected, ICEConnectionStateFailed: |
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if rtc.Stream != nil { |
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rtc.Stream.Close() |
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} |
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} |
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} |
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}) |
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}) |
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rtc.PeerConnection = peerConnection |
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rtc.PeerConnection = peerConnection |
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if rtc.RTP.Publish(streamPath) { |
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if rtc.RTP.Publish(streamPath) { |
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//f, _ := os.OpenFile("resource/live/rtc.h264", os.O_TRUNC|os.O_WRONLY, 0666)
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var h264 codecs.H264Packet |
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rtc.Stream.Type = "WebRTC" |
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rtc.Stream.Type = "WebRTC" |
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peerConnection.OnTrack(func(track *Track, receiver *RTPReceiver) { |
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peerConnection.OnTrack(func(track *Track, receiver *RTPReceiver) { |
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defer rtc.Stream.Close() |
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defer rtc.Stream.Close() |
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@ -113,6 +280,8 @@ func (rtc *WebRTC) Publish(streamPath string) bool { |
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if err = pack.Unmarshal(b[:i]); err != nil { |
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if err = pack.Unmarshal(b[:i]); err != nil { |
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return |
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return |
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} |
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} |
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h264.Unmarshal(pack.Payload) |
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// f.Write(bytes)
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} |
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} |
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}) |
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}) |
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} else { |
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} else { |
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@ -120,8 +289,32 @@ func (rtc *WebRTC) Publish(streamPath string) bool { |
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} |
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} |
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return true |
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return true |
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} |
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} |
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func (rtc *WebRTC) GetAnswer(offer SessionDescription) ([]byte, error) { |
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if err := rtc.SetRemoteDescription(offer); err != nil { |
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Println(err) |
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return nil, err |
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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 nil, err |
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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 nil, err |
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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 bytes, err |
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} else { |
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return bytes, nil |
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} |
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} |
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func run() { |
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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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http.HandleFunc("/webrtc/play", func(w http.ResponseWriter, r *http.Request) { |
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streamPath := r.URL.Query().Get("streamPath") |
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streamPath := r.URL.Query().Get("streamPath") |
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offer := SessionDescription{} |
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offer := SessionDescription{} |
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bytes, err := ioutil.ReadAll(r.Body) |
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bytes, err := ioutil.ReadAll(r.Body) |
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@ -131,30 +324,38 @@ func run() { |
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return |
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return |
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} |
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} |
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rtc := new(WebRTC) |
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rtc := new(WebRTC) |
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if rtc.Publish(streamPath) { |
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rtc.RemoteAddr = r.RemoteAddr |
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// Set the remote SessionDescription
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if rtc.Play(streamPath) { |
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if err = rtc.SetRemoteDescription(offer); err != nil { |
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if bytes, err = rtc.GetAnswer(offer); err == nil { |
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w.Write(bytes) |
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} else { |
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Println(err) |
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Println(err) |
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w.Write([]byte(err.Error())) |
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return |
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return |
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} |
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} |
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} else { |
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// Create answer
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w.Write([]byte(`{"errmsg":"bad name"}`)) |
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answer, err := rtc.CreateAnswer(nil) |
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} |
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if err != nil { |
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}) |
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Println(err) |
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http.HandleFunc("/webrtc/publish", func(w http.ResponseWriter, r *http.Request) { |
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return |
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streamPath := r.URL.Query().Get("streamPath") |
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} |
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offer := SessionDescription{} |
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bytes, err := ioutil.ReadAll(r.Body) |
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// Sets the LocalDescription, and starts our UDP listeners
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err = json.Unmarshal(bytes, &offer) |
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if err = rtc.SetLocalDescription(answer); err != nil { |
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if err != nil { |
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Println(err) |
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Println(err) |
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return |
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return |
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} |
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} |
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if bytes, err = json.Marshal(answer); err != nil { |
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rtc := new(WebRTC) |
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rtc.RemoteAddr = r.RemoteAddr |
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if rtc.Publish(streamPath) { |
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if bytes, err = rtc.GetAnswer(offer); err == nil { |
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w.Write(bytes) |
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} else { |
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Println(err) |
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Println(err) |
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w.Write([]byte(err.Error())) |
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return |
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return |
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} |
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} |
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w.Write(bytes) |
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} else { |
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} else { |
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w.Write([]byte(`{"errmsg":"bad name"}`)) |
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w.Write([]byte(`{"errmsg":"bad name"}`)) |
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} |
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} |
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langhuihui
5 years ago