mirror of
https://github.com/cnotch/ipchub.git
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462 lines
10 KiB
Go
Executable File
462 lines
10 KiB
Go
Executable File
// Copyright (c) 2019,CAOHONGJU All rights reserved.
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// Use of this source code is governed by a MIT-style
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// license that can be found in the LICENSE file.
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package wsp
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import (
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"bufio"
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"bytes"
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"fmt"
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"io"
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"net/url"
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"runtime/debug"
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"strings"
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"sync"
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"time"
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"github.com/cnotch/ipchub/config"
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"github.com/cnotch/ipchub/media"
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"github.com/cnotch/ipchub/network/websocket"
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"github.com/cnotch/ipchub/provider/security"
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"github.com/cnotch/ipchub/service/rtsp"
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"github.com/cnotch/ipchub/stats"
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"github.com/cnotch/xlog"
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"github.com/pixelbender/go-sdp/sdp"
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)
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const (
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statusInit = iota
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statusReady
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statusPlaying
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)
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const (
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rtspURLPrefix = "rtsp://" // RTSP地址前缀
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)
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// Session RTSP 会话
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type Session struct {
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// 创建时设置
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svr *Server
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channelID string
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logger *xlog.Logger
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closed bool
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paused bool
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lsession string // 本地会话标识
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timeout time.Duration
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conn websocket.Conn
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lockW sync.Mutex
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dataChannel websocket.Conn
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// DESCRIBE 后设置
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url *url.URL
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path string
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rawSdp string
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sdp *sdp.Session
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aControl string
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vControl string
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aCodec string
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vCodec string
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// Setup 后设置
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transport rtsp.RTPTransport
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// 启动流媒体传输后设置
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status int // session状态
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source *media.Stream
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cid *media.CID
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}
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func newSession(svr *Server, conn websocket.Conn, channelID string) *Session {
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session := &Session{
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svr: svr,
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channelID: channelID,
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lsession: security.NewID().Base64(),
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timeout: config.NetTimeout() * time.Duration(2),
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conn: conn,
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transport: rtsp.RTPTransport{
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Mode: rtsp.PlaySession, // 默认为播放
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Type: rtsp.RTPUnknownTrans,
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},
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status: statusInit,
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}
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for i := 0; i < 4; i++ {
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session.transport.Channels[i] = -1
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session.transport.ClientPorts[i] = -1
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}
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session.logger = svr.logger.With(xlog.Fields(
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xlog.F("channel", channelID),
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xlog.F("path", conn.Path())))
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return session
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}
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// 设置rtp数据通道
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func (s *Session) setDataChannel(dc websocket.Conn) {
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s.lockW.Lock()
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s.dataChannel = dc
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s.lockW.Unlock()
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}
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// Addr Session地址
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func (s *Session) Addr() string {
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return s.conn.RemoteAddr().String()
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}
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// Consume 消费媒体包
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func (s *Session) Consume(p Pack) {
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if s.closed || s.paused {
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return
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}
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buf := buffers.Get().(*bytes.Buffer)
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buf.Reset()
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defer buffers.Put(buf)
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p2 := p.(*rtsp.RTPPack)
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p2.Write(buf, s.transport.Channels[:])
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var err error
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s.lockW.Lock()
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if s.dataChannel != nil {
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_, err = s.dataChannel.Write(buf.Bytes())
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}
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s.lockW.Unlock()
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if err != nil {
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s.logger.Errorf("send pack error = %v , close socket", err)
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s.Close()
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return
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}
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}
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// Close 关闭会话
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func (s *Session) Close() error {
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if s.closed {
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return nil
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}
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s.closed = true
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s.paused = false
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s.conn.Close()
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s.lockW.Lock()
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if s.dataChannel != nil {
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s.dataChannel.Close()
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}
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s.lockW.Unlock()
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return nil
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}
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func (s *Session) process() {
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var err error
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defer func() {
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if r := recover(); r != nil {
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s.logger.Errorf("wsp channel panic, %v \n %s", r, debug.Stack())
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}
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if err != nil {
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if err == io.EOF { // 如果客户端断开提醒
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s.logger.Warn("websocket disconnect actively")
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} else if !s.closed { // 如果主动关闭，不提示
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s.logger.Error(err.Error())
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}
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}
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// 删除通道
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s.svr.sessions.Delete(s.channelID)
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// 停止消费
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if s.cid != nil {
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s.source.StopConsume(*s.cid)
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s.cid = nil
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s.source = nil
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}
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// 关闭连接
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s.Close()
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// 重置到初始状态
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s.conn = nil
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s.dataChannel = nil
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s.status = statusInit
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stats.WspConns.Release()
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s.logger.Info("close wsp channel")
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}()
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s.logger.Info("open wsp channel")
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stats.WspConns.Add() // 增加一个 RTSP 连接计数
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for !s.closed {
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deadLine := time.Time{}
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if s.timeout > 0 {
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deadLine = time.Now().Add(s.timeout)
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}
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if err = s.conn.SetReadDeadline(deadLine); err != nil {
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break
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}
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var req *Request
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req, err = DecodeRequest(s.conn, s.logger)
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if err != nil {
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break
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}
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if req.Cmd != CmdWrap {
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s.logger.Error("must is WRAP command request")
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break
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}
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// 从包装命令中提取 rtsp 请求
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var rtspReq *rtsp.Request
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rtspReq, err = rtsp.ReadRequest(bufio.NewReader(bytes.NewBufferString(req.Body)))
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if err != nil {
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break
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}
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// 处理请求，并获得响应
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rtspResp := s.onRequest(rtspReq)
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// 发送响应
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buf := buffers.Get().(*bytes.Buffer)
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buf.Reset()
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defer buffers.Put(buf)
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req.ResponseOK(buf, map[string]string{FieldChannel: s.channelID}, "")
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rtspResp.Write(buf)
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_, err = s.conn.Write(buf.Bytes())
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if err != nil {
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break
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}
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s.logger.Debugf("wsp ===>>>\r\n%s", buf.String())
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// 关闭通道
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if rtspReq.Method == rtsp.MethodTeardown {
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break
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}
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}
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}
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func (s *Session) onRequest(req *rtsp.Request) *rtsp.Response {
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resp := s.newResponse(rtsp.StatusOK, req)
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// 预处理
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continueProcess := s.onPreprocess(resp, req)
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if !continueProcess {
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return resp
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}
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switch req.Method {
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case rtsp.MethodDescribe:
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s.onDescribe(resp, req)
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case rtsp.MethodSetup:
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s.onSetup(resp, req)
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case rtsp.MethodPlay:
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s.onPlay(resp, req)
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case rtsp.MethodPause:
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s.onPause(resp, req)
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default:
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// 状态不支持的方法
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resp.StatusCode = rtsp.StatusMethodNotValidInThisState
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}
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return resp
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}
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func (s *Session) onDescribe(resp *rtsp.Response, req *rtsp.Request) {
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// TODO: 检查 accept 中的类型是否包含 sdp
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s.url = req.URL
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s.path = s.conn.Path() // 使用websocket路径
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// s.path = utils.CanonicalPath(req.URL.Path)
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stream := media.GetOrCreate(s.path)
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if stream == nil {
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resp.StatusCode = rtsp.StatusNotFound
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return
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}
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// 从流中取 sdp
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sdpRaw := stream.Sdp()
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if len(sdpRaw) == 0 {
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resp.StatusCode = rtsp.StatusNotFound
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return
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}
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err := s.parseSdp(sdpRaw)
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if err != nil { // TODO：需要更好的处理方式
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resp.StatusCode = rtsp.StatusNotFound
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return
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}
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resp.Header.Set(rtsp.FieldContentType, "application/sdp")
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resp.Body = s.rawSdp
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}
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func (s *Session) onSetup(resp *rtsp.Response, req *rtsp.Request) {
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// a=control:streamid=1
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// a=control:rtsp://192.168.1.165/trackID=1
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// a=control:?ctype=video
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setupURL := &url.URL{}
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*setupURL = *req.URL
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if setupURL.Port() == "" {
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setupURL.Host = fmt.Sprintf("%s:554", setupURL.Host)
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}
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setupPath := setupURL.String()
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//setupPath = setupPath[strings.LastIndex(setupPath, "/")+1:]
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vPath := getControlPath(s.vControl)
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if vPath == "" {
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resp.StatusCode = rtsp.StatusInternalServerError
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resp.Status = "Invalid VControl"
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return
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}
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aPath := getControlPath(s.aControl)
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ts := req.Header.Get(rtsp.FieldTransport)
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resp.Header.Set(rtsp.FieldTransport, ts) // 先回写transport
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// 检查控制路径
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chindex := -1
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if setupPath == aPath || (aPath != "" && strings.LastIndex(setupPath, aPath) == len(setupPath)-len(aPath)) {
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chindex = int(rtsp.ChannelAudio)
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} else if setupPath == vPath || (vPath != "" && strings.LastIndex(setupPath, vPath) == len(setupPath)-len(vPath)) {
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chindex = int(rtsp.ChannelVideo)
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} else { // 找不到被 Setup 的资源
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resp.StatusCode = rtsp.StatusInternalServerError
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resp.Status = fmt.Sprintf("SETUP Unkown control:%s", setupPath)
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return
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}
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err := s.transport.ParseTransport(chindex, ts)
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if err != nil {
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resp.StatusCode = rtsp.StatusInvalidParameter
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resp.Status = err.Error()
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return
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}
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// 检查必须是play模式
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if rtsp.PlaySession != s.transport.Mode {
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resp.StatusCode = rtsp.StatusInvalidParameter
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resp.Status = "can't setup as record"
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return
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}
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if s.transport.Type != rtsp.RTPTCPUnicast { // 需要修改回复的transport
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resp.StatusCode = rtsp.StatusUnsupportedTransport
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resp.Status = "websocket only support tcp unicast"
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return
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}
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if s.status < statusReady { // 初始状态切换到Ready
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s.status = statusReady
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}
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}
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func (s *Session) onPlay(resp *rtsp.Response, req *rtsp.Request) {
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if s.status == statusPlaying {
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s.paused = false
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return
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}
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stream := media.GetOrCreate(s.path)
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if stream == nil {
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resp.StatusCode = rtsp.StatusNotFound
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return
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}
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resp.Header.Set(rtsp.FieldRange, req.Header.Get(rtsp.FieldRange))
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if s.cid == nil {
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s.source = stream
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// cid := stream.StartConsume(s)
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cid := stream.StartConsumeNoGopCache(s, media.RTPPacket, "wsp")
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s.cid = &cid
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}
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s.status = statusPlaying
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s.paused = false
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return
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}
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func (s *Session) onPause(resp *rtsp.Response, req *rtsp.Request) {
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if s.status == statusPlaying {
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s.paused = true
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}
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}
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func (s *Session) onPreprocess(resp *rtsp.Response, req *rtsp.Request) (continueProcess bool) {
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// Options
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if req.Method == rtsp.MethodOptions {
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resp.Header.Set(rtsp.FieldPublic, "DESCRIBE, SETUP, TEARDOWN, PLAY, OPTIONS, ANNOUNCE")
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return false
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}
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// 关闭请求
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if req.Method == rtsp.MethodTeardown {
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return false
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}
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// 检查状态下的方法
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switch s.status {
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case statusReady:
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continueProcess = req.Method == rtsp.MethodSetup ||
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req.Method == rtsp.MethodPlay
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case statusPlaying:
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continueProcess = (req.Method == rtsp.MethodPlay ||
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req.Method == rtsp.MethodPause)
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default:
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continueProcess = !(req.Method == rtsp.MethodPlay ||
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req.Method == rtsp.MethodRecord)
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}
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if !continueProcess {
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resp.StatusCode = rtsp.StatusMethodNotValidInThisState
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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 (s *Session) newResponse(code int, req *rtsp.Request) *rtsp.Response {
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resp := &rtsp.Response{
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StatusCode: code,
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Header: make(rtsp.Header),
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Request: req,
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}
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resp.Header.Set(rtsp.FieldCSeq, req.Header.Get(rtsp.FieldCSeq))
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resp.Header.Set(rtsp.FieldSession, s.lsession)
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return resp
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}
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func (s *Session) parseSdp(rawSdp string) (err error) {
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// 从流中取 sdp
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s.rawSdp = rawSdp
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// 解析
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s.sdp, err = sdp.ParseString(s.rawSdp)
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if err != nil {
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return
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}
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for _, media := range s.sdp.Media {
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switch media.Type {
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case "video":
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s.vControl = media.Attributes.Get("control")
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s.vCodec = media.Format[0].Name
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case "audio":
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s.aControl = media.Attributes.Get("control")
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s.aCodec = media.Format[0].Name
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}
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}
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return
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}
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func getControlPath(ctrl string) (path string) {
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if len(ctrl) >= len(rtspURLPrefix) && strings.EqualFold(ctrl[:len(rtspURLPrefix)], rtspURLPrefix) {
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ctrlURL, err := url.Parse(ctrl)
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if err != nil {
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return
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}
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if ctrlURL.Port() == "" {
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ctrlURL.Host = fmt.Sprintf("%s:554", ctrlURL.Hostname())
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}
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return ctrlURL.String()
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}
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return ctrl
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}
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