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https://github.com/aler9/gortsplib
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* switch from v4 to v5 * remove deprecated entities * remove "2" suffix from entities * rename TransportProtocol into Protocol
168 lines
3.9 KiB
Go
168 lines
3.9 KiB
Go
// Package main contains an example.
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package main
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import (
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"log"
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"sync"
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"github.com/bluenviron/gortsplib/v5"
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"github.com/bluenviron/gortsplib/v5/pkg/base"
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"github.com/bluenviron/gortsplib/v5/pkg/description"
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"github.com/bluenviron/gortsplib/v5/pkg/format"
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"github.com/pion/rtp"
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)
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// This example shows how to:
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// 1. create a RTSP server which accepts plain connections.
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// 2. create a stream with an audio direct channel and an audio back channel.
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// 3. write the audio direct channel to readers, read the back channel from readers.
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type serverHandler struct {
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server *gortsplib.Server
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stream *gortsplib.ServerStream
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mutex sync.RWMutex
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}
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// called when a connection is opened.
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func (sh *serverHandler) OnConnOpen(_ *gortsplib.ServerHandlerOnConnOpenCtx) {
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log.Printf("conn opened")
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}
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// called when a connection is closed.
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func (sh *serverHandler) OnConnClose(ctx *gortsplib.ServerHandlerOnConnCloseCtx) {
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log.Printf("conn closed (%v)", ctx.Error)
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}
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// called when a session is opened.
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func (sh *serverHandler) OnSessionOpen(_ *gortsplib.ServerHandlerOnSessionOpenCtx) {
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log.Printf("session opened")
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}
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// called when a session is closed.
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func (sh *serverHandler) OnSessionClose(_ *gortsplib.ServerHandlerOnSessionCloseCtx) {
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log.Printf("session closed")
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}
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// called when receiving a DESCRIBE request.
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func (sh *serverHandler) OnDescribe(
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_ *gortsplib.ServerHandlerOnDescribeCtx,
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) (*base.Response, *gortsplib.ServerStream, error) {
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log.Printf("DESCRIBE request")
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sh.mutex.RLock()
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defer sh.mutex.RUnlock()
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return &base.Response{
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StatusCode: base.StatusOK,
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}, sh.stream, nil
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}
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// called when receiving a SETUP request.
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func (sh *serverHandler) OnSetup(
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_ *gortsplib.ServerHandlerOnSetupCtx,
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) (*base.Response, *gortsplib.ServerStream, error) {
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log.Printf("SETUP request")
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sh.mutex.RLock()
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defer sh.mutex.RUnlock()
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return &base.Response{
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StatusCode: base.StatusOK,
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}, sh.stream, nil
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}
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// called when receiving a PLAY request.
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func (sh *serverHandler) OnPlay(ctx *gortsplib.ServerHandlerOnPlayCtx) (*base.Response, error) {
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log.Printf("PLAY request")
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// called when receiving a RTP packet
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ctx.Session.OnPacketRTPAny(func(m *description.Media, _ format.Format, pkt *rtp.Packet) {
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// decode timestamp
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pts, ok := ctx.Session.PacketPTS(m, pkt)
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if !ok {
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return
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}
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log.Printf("incoming RTP packet with PTS=%v size=%v", pts, len(pkt.Payload))
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})
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return &base.Response{
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StatusCode: base.StatusOK,
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}, nil
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}
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func main() {
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h := &serverHandler{}
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// prevent clients from connecting to the server until the stream is properly set up
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h.mutex.Lock()
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// create the server
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h.server = &gortsplib.Server{
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Handler: h,
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RTSPAddress: ":8554",
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UDPRTPAddress: ":8000",
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UDPRTCPAddress: ":8001",
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MulticastIPRange: "224.1.0.0/16",
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MulticastRTPPort: 8002,
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MulticastRTCPPort: 8003,
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}
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// start the server
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err := h.server.Start()
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if err != nil {
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panic(err)
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}
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defer h.server.Close()
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// create a RTSP description
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desc := &description.Session{
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Medias: []*description.Media{
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// direct channel
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{
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Type: description.MediaTypeAudio,
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Formats: []format.Format{&format.G711{
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PayloadTyp: 8,
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MULaw: false,
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SampleRate: 8000,
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ChannelCount: 1,
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}},
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},
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// back channel
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{
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Type: description.MediaTypeAudio,
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IsBackChannel: true,
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Formats: []format.Format{&format.G711{
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PayloadTyp: 8,
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MULaw: false,
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SampleRate: 8000,
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ChannelCount: 1,
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}},
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},
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},
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}
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// create a server stream
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h.stream = &gortsplib.ServerStream{
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Server: h.server,
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Desc: desc,
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}
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err = h.stream.Initialize()
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if err != nil {
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panic(err)
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}
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defer h.stream.Close()
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// create audio streamer
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r := &audioStreamer{stream: h.stream}
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r.initialize()
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defer r.close()
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// allow clients to connect
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h.mutex.Unlock()
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// wait until a fatal error
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log.Printf("server is ready on %s", h.server.RTSPAddress)
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panic(h.server.Wait())
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}
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