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支持rtsp over tcp

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This commit is contained in:
yangjiechina
2024-03-17 20:20:43 +08:00
parent d325daeb8b
commit 7694237dc7
5 changed files with 102 additions and 56 deletions
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11
rtsp/rtp_track.go
View File

@@ -19,12 +19,13 @@ type rtpTrack struct {
tmp [][]byte
}
func NewRTPTrack(muxer librtp.Muxer, pt byte, rate int) *rtpTrack {
func NewRTPTrack(muxer librtp.Muxer, pt byte, rate int, mediaType utils.AVMediaType) *rtpTrack {
stream := &rtpTrack{
pt: pt,
rate: rate,
muxer: muxer,
buffer: make([]byte, 1500),
pt: pt,
rate: rate,
muxer: muxer,
buffer: make([]byte, 1500),
mediaType: mediaType,
}
return stream

61
rtsp/rtsp_session.go
View File

@@ -15,6 +15,8 @@ import (
"time"
)
type SessionState int
const (
MethodOptions = "OPTIONS"
MethodDescribe = "DESCRIBE"
@@ -29,6 +31,13 @@ const (
MethodRecord = "RECORD"
Version = "RTSP/1.0"
SessionSateOptions = SessionState(0x1)
SessionSateDescribe = SessionState(0x2)
SessionSateSetup = SessionState(0x3)
SessionSatePlay = SessionState(0x4)
SessionSateTeardown = SessionState(0x5)
SessionSatePause = SessionState(0x6)
)
type requestHandler interface {
@@ -181,27 +190,29 @@ func (s *session) onSetup(sourceId string, index int, headers textproto.MIMEHead
var clientRtpPort int
var clientRtcpPort int
tcp := "RTP/AVP" != split[0] && "RTP/AVP/UDP" != split[0]
for _, value := range split {
if !strings.HasPrefix(value, "client_port=") {
continue
}
if !tcp {
for _, value := range split {
if !strings.HasPrefix(value, "client_port=") {
continue
}
pairPort := strings.Split(value[len("client_port="):], "-")
if len(pairPort) != 2 {
return fmt.Errorf("failed to parsing client_port:%s", value)
}
pairPort := strings.Split(value[len("client_port="):], "-")
if len(pairPort) != 2 {
return fmt.Errorf("failed to parsing client_port:%s", value)
}
port, err := strconv.Atoi(pairPort[0])
if err != nil {
return err
}
clientRtpPort = port
port, err := strconv.Atoi(pairPort[0])
if err != nil {
return err
}
clientRtpPort = port
port, err = strconv.Atoi(pairPort[1])
if err != nil {
return err
port, err = strconv.Atoi(pairPort[1])
if err != nil {
return err
}
clientRtcpPort = port
}
clientRtcpPort = port
}
rtpPort, rtcpPort, err := s.sink_.addTrack(index, tcp)
@@ -211,7 +222,15 @@ func (s *session) onSetup(sourceId string, index int, headers textproto.MIMEHead
println(clientRtpPort)
println(clientRtcpPort)
responseHeader := transportHeader + ";server_port=" + fmt.Sprintf("%d-%d", rtpPort, rtcpPort) + ";ssrc=FFFFFFFF"
responseHeader := transportHeader
if tcp {
//修改interleaved为实际的stream index
responseHeader += ";interleaved=" + fmt.Sprintf("%d-%d", index, index)
} else {
responseHeader += ";server_port=" + fmt.Sprintf("%d-%d", rtpPort, rtcpPort)
}
responseHeader += ";ssrc=FFFFFFFF"
response := NewOKResponse(headers.Get("Cseq"))
response.Header.Set("Transport", responseHeader)
response.Header.Set("Session", s.sessionId)
@@ -226,7 +245,11 @@ func (s *session) onPlay(sourceId string, headers textproto.MIMEHeader) error {
response.Header.Set("Session", sessionHeader)
}
return s.response(response, nil)
err := s.response(response, nil)
if err == nil {
s.sink_.playing = true
}
return err
}
func (s *session) onTeardown() {

32
rtsp/rtsp_sink.go
View File

@@ -15,7 +15,10 @@ type sink struct {
//一个rtsp源，可能存在多个流, 每个流都需要拉取拉取
tracks []*rtspTrack
sdpCB func(sdp string)
sdpCb func(sdp string)
tcp bool
playing bool
}
func NewSink(id stream.SinkId, sourceId string, conn net.Conn, cb func(sdp string)) stream.ISink {
@@ -23,6 +26,8 @@ func NewSink(id stream.SinkId, sourceId string, conn net.Conn, cb func(sdp strin
stream.SinkImpl{Id_: id, SourceId_: sourceId, Protocol_: stream.ProtocolRtsp, Conn: conn},
nil,
cb,
false,
false,
}
}
@@ -40,18 +45,7 @@ func (s *sink) addTrack(index int, tcp bool) (int, int, error) {
track := rtspTrack{}
if tcp {
err = rtspTransportManger.AllocTransport(true, func(port int) {
var addr *net.TCPAddr
addr, err = net.ResolveTCPAddr("tcp", fmt.Sprintf("%s:%d", "0.0.0.0", port))
if err == nil {
track.rtp = &transport.TCPServer{}
track.rtp.SetHandler2(track.onTCPConnected, nil, track.onTCPDisconnected)
err = track.rtp.Bind(addr)
}
rtpPort = port
})
s.tcp = true
} else {
err = rtspTransportManger.AllocPairTransport(func(port int) {
//rtp port
@@ -91,16 +85,18 @@ func (s *sink) addTrack(index int, tcp bool) (int, int, error) {
func (s *sink) input(index int, data []byte) error {
utils.Assert(index < cap(s.tracks))
//拉流方还没有连上来
s.tracks[index].pktCount++
s.tracks[index].rtpConn.Write(data)
if s.tcp {
s.Conn.Write(data)
} else {
s.tracks[index].rtpConn.Write(data)
}
return nil
}
func (s *sink) isConnected(index int) bool {
return s.tracks[index] != nil && s.tracks[index].rtpConn != nil
return s.playing && (s.tcp || (s.tracks[index] != nil && s.tracks[index].rtpConn != nil))
}
func (s *sink) pktCount(index int) int {
@@ -109,7 +105,7 @@ func (s *sink) pktCount(index int) int {
// SendHeader 回调rtsp流的sdp信息
func (s *sink) SendHeader(data []byte) error {
s.sdpCB(string(data))
s.sdpCb(string(data))
return nil
}

53
rtsp/rtsp_stream.go
View File

@@ -10,6 +10,11 @@ import (
"net"
)
const (
OverTcpHeaderSize = 4
OverTcpMagic = 0x24
)
// 低延迟是rtsp特性, 不考虑实现GOP缓存
type tranStream struct {
stream.TransStreamImpl
@@ -38,39 +43,59 @@ func NewTransStream(addr net.IPAddr, urlFormat string) stream.ITransStream {
}
func (t *tranStream) onAllocBuffer(params interface{}) []byte {
return t.rtpTracks[params.(int)].buffer
return t.rtpTracks[params.(int)].buffer[OverTcpHeaderSize:]
}
func (t *tranStream) onRtpPacket(data []byte, timestamp uint32, params interface{}) {
index := params.(int)
track := t.rtpTracks[index]
if t.rtpTracks[index].cache && t.rtpTracks[index].header == nil {
bytes := make([]byte, len(data))
copy(bytes, data)
if track.cache && track.header == nil {
bytes := make([]byte, OverTcpHeaderSize+len(data))
copy(bytes[OverTcpHeaderSize:], data)
t.rtpTracks[index].tmp = append(t.rtpTracks[index].tmp, bytes)
track.tmp = append(track.tmp, bytes)
t.overTCP(bytes, index)
return
}
for _, iSink := range t.Sinks {
if !iSink.(*sink).isConnected(index) {
for _, value := range t.Sinks {
sink_ := value.(*sink)
if !sink_.isConnected(index) {
continue
}
if iSink.(*sink).pktCount(index) < 1 && utils.AVMediaTypeVideo == t.rtpTracks[index].mediaType {
if sink_.pktCount(index) < 1 && utils.AVMediaTypeVideo == track.mediaType {
seq := binary.BigEndian.Uint16(data[2:])
count := len(t.rtpTracks[index].header)
count := len(track.header)
for i, rtp := range t.rtpTracks[index].header {
librtp.RollbackSeq(rtp, int(seq)-(count-i-1))
iSink.(*sink).input(index, rtp)
for i, rtp := range track.header {
librtp.RollbackSeq(rtp[OverTcpHeaderSize:], int(seq)-(count-i-1))
if sink_.tcp {
sink_.input(index, rtp)
} else {
sink_.input(index, rtp[OverTcpHeaderSize:])
}
}
}
iSink.(*sink).input(index, data)
end := OverTcpHeaderSize + len(data)
t.overTCP(track.buffer[:end], index)
if sink_.tcp {
sink_.input(index, track.buffer[:end])
} else {
sink_.input(index, data)
}
}
}
func (t *tranStream) overTCP(data []byte, channel int) {
data[0] = OverTcpMagic
data[1] = byte(channel)
binary.BigEndian.PutUint16(data[2:], uint16(len(data)-4))
}
func (t *tranStream) Input(packet utils.AVPacket) error {
stream_ := t.rtpTracks[packet.Index()]
if utils.AVMediaTypeAudio == packet.MediaType() {
@@ -135,7 +160,7 @@ func (t *tranStream) AddTrack(stream utils.AVStream) error {
muxer.SetAllocHandler(t.onAllocBuffer)
muxer.SetWriteHandler(t.onRtpPacket)
t.rtpTracks = append(t.rtpTracks, NewRTPTrack(muxer, byte(payloadType.Pt), payloadType.ClockRate))
t.rtpTracks = append(t.rtpTracks, NewRTPTrack(muxer, byte(payloadType.Pt), payloadType.ClockRate, stream.Type()))
muxer.SetParams(len(t.rtpTracks) - 1)
return nil
}