feat: 支持国标语音广播

2025-09-26 19:21:14 +08:00 · 2025-05-09 20:56:17 +08:00
parent 9f22072406
commit b254fff76c
19 changed files with 1196 additions and 271 deletions
--- a/api.go
+++ b/api.go
@@ -104,19 +104,18 @@ func startApiServer(addr string) {
 	apiServer.router.HandleFunc("/api/v1/streams/statistics", nil) // 统计所有推拉流

 	if stream.AppConfig.GB28181.Enable {
-		apiServer.router.HandleFunc("/api/v1/gb28181/forward", filterRequestBodyParams(apiServer.OnGBSourceForward, &GBForwardParams{}))     // 设置级联转发目标，停止级联调用sink/close接口，级联断开会走on_play_done事件通知
-		apiServer.router.HandleFunc("/api/v1/gb28181/source/create", filterRequestBodyParams(apiServer.OnGBSourceCreate, &GBSourceParams{})) // 创建国标推流源
-		apiServer.router.HandleFunc("/api/v1/gb28181/source/connect", filterRequestBodyParams(apiServer.OnGBSourceConnect, &GBConnect{}))    // 为国标TCP主动推流，设置连接地址
+		apiServer.router.HandleFunc("/ws/v1/gb28181/talk", apiServer.OnGBTalk) // 对讲的主讲人WebSocket连接
+		apiServer.router.HandleFunc("/api/v1/gb28181/offer/create", filterRequestBodyParams(apiServer.OnGBOfferCreate, &SourceSDP{}))
+		apiServer.router.HandleFunc("/api/v1/gb28181/answer/create", filterRequestBodyParams(apiServer.OnGBAnswerCreate, &GBOffer{}))
+		apiServer.router.HandleFunc("/api/v1/gb28181/answer/set", filterRequestBodyParams(apiServer.OnGBSourceConnect, &SourceSDP{})) // active拉流模式下, 设置对方的地址
 	}

 	apiServer.router.HandleFunc("/api/v1/gc/force", func(writer http.ResponseWriter, request *http.Request) {
 		runtime.GC()
-		writer.WriteHeader(http.StatusOK)
 	})

-	apiServer.router.HandleFunc("/rtc.html", func(writer http.ResponseWriter, request *http.Request) {
-		http.ServeFile(writer, request, "./rtc.html")
-	})
+	apiServer.router.PathPrefix("/web/").Handler(http.StripPrefix("/web/", http.FileServer(http.Dir("./web"))))
+
 	http.Handle("/", apiServer.router)

 	srv := &http.Server{
--- a/api_gb.go
+++ b/api_gb.go
@@ -2,6 +2,7 @@ package main

 import (
 	"fmt"
+	"github.com/lkmio/avformat/bufio"
 	"github.com/lkmio/avformat/utils"
 	"github.com/lkmio/lkm/gb28181"
 	"github.com/lkmio/lkm/log"
@@ -9,47 +10,41 @@ import (
 	"net"
 	"net/http"
 	"strconv"
-	"strings"
 )

 const (
-	InviteTypeLive = iota
-	InviteTypePlayback
-	InviteTypeDownload
-	InviteTypeBroadcast
-	InviteTypeTalk
+	InviteTypePlay      = "play"
+	InviteTypePlayback  = "playback"
+	InviteTypeDownload  = "download"
+	InviteTypeBroadcast = "broadcast"
+	InviteTypeTalk      = "talk"
 )

-type GBForwardParams struct {
-	Source     string `json:"source"` // GetSourceID
-	Addr       string `json:"addr"`
-	SSRC       uint32 `json:"ssrc"`
-	OfferSetup string `json:"offer_setup"`
-	Setup      string `json:"setup"`
-	Type       int    `json:"type"` // live/download/playback/talk/broadcast
+type SDP struct {
+	SessionName string `json:"session_name,omitempty"` // play/download/playback/talk/broadcast
+	Addr        string `json:"addr,omitempty"`         // 连接地址
+	SSRC        string `json:"ssrc,omitempty"`
+	Setup       string `json:"setup,omitempty"`     // active/passive
+	Transport   string `json:"transport,omitempty"` // tcp/udp
 }

-type GBSourceParams struct {
+type SourceSDP struct {
 	Source string `json:"source"` // GetSourceID
-	Setup  string `json:"setup"`  // active/passive
-	SSRC   uint32 `json:"ssrc,omitempty"`
-	Type   int    `json:"type"` // live/download/playback/talk/broadcast
+	SDP
 }

-type GBConnect struct {
-	Source     string `json:"source"` // GetSourceID
-	RemoteAddr string `json:"remote_addr"`
+type GBOffer struct {
+	SourceSDP
+	AnswerSetup string `json:"answer_setup,omitempty"` // 希望应答的连接方式
 }

-func (api *ApiServer) OnGBSourceCreate(v *GBSourceParams, w http.ResponseWriter, r *http.Request) {
+func (api *ApiServer) OnGBSourceCreate(v *SourceSDP, w http.ResponseWriter, r *http.Request) {
 	log.Sugar.Infof("创建国标源: %v", v)

 	// 返回收流地址
 	response := &struct {
-		IP   string   `json:"ip"`
-		Port int      `json:"port,omitempty"`
+		SDP
 		Urls []string `json:"urls"`
-		SSRC string   `json:"ssrc,omitempty"`
 	}{}

 	var err error
@@ -92,7 +87,7 @@ func (api *ApiServer) OnGBSourceCreate(v *GBSourceParams, w http.ResponseWriter,
 	}

 	var ssrc string
-	if v.Type == InviteTypeDownload || v.Type == InviteTypePlayback {
+	if v.SessionName == InviteTypeDownload || v.SessionName == InviteTypePlayback {
 		ssrc = gb28181.GetVodSSRC()
 	} else {
 		ssrc = gb28181.GetLiveSSRC()
@@ -104,14 +99,13 @@ func (api *ApiServer) OnGBSourceCreate(v *GBSourceParams, w http.ResponseWriter,
 		return
 	}

-	response.IP = stream.AppConfig.PublicIP
-	response.Port = port
+	response.Addr = net.JoinHostPort(stream.AppConfig.PublicIP, strconv.Itoa(port))
 	response.Urls = stream.GetStreamPlayUrls(v.Source)
 	response.SSRC = ssrc
 	httpResponseOK(w, response)
 }

-func (api *ApiServer) OnGBSourceConnect(v *GBConnect, w http.ResponseWriter, r *http.Request) {
+func (api *ApiServer) OnGBSourceConnect(v *SourceSDP, w http.ResponseWriter, r *http.Request) {
 	log.Sugar.Infof("设置国标主动拉流连接地址: %v", v)

 	var err error
@@ -135,7 +129,7 @@ func (api *ApiServer) OnGBSourceConnect(v *GBConnect, w http.ResponseWriter, r *
 		return
 	}

-	addr, err := net.ResolveTCPAddr("tcp", v.RemoteAddr)
+	addr, err := net.ResolveTCPAddr("tcp", v.Addr)
 	if err != nil {
 		return
 	}
@@ -145,62 +139,133 @@ func (api *ApiServer) OnGBSourceConnect(v *GBConnect, w http.ResponseWriter, r *
 	}
 }

-func (api *ApiServer) OnGBSourceForward(v *GBForwardParams, w http.ResponseWriter, r *http.Request) {
-	log.Sugar.Infof("设置国标级联转发: %v", v)
+func (api *ApiServer) OnGBOfferCreate(v *SourceSDP, w http.ResponseWriter, r *http.Request) {
+	// 预览下级设备
+	if v.SessionName == "" || v.SessionName == InviteTypePlay ||
+		v.SessionName == InviteTypePlayback ||
+		v.SessionName == InviteTypeDownload {
+		api.OnGBSourceCreate(v, w, r)
+	} else {
+		// 向上级转发广播和对讲, 或者是向设备发送invite talk
+	}
+}

+func (api *ApiServer) OnGBAnswerCreate(v *GBOffer, w http.ResponseWriter, r *http.Request) {
+	log.Sugar.Infof("创建应答 offer: %v", v)
+
+	var sink stream.Sink
 	var err error
 	// 响应错误消息
 	defer func() {
 		if err != nil {
-			log.Sugar.Errorf("设置级联转发失败 err: %s", err.Error())
+			log.Sugar.Errorf("创建应答失败 err: %s", err.Error())
 			httpResponseError(w, err.Error())
+
+			if sink != nil {
+				sink.Close()
+			}
 		}
 	}()

 	source := stream.SourceManager.Find(v.Source)
 	if source == nil {
 		err = fmt.Errorf("%s 源不存在", v.Source)
-	} else if source.GetType() != stream.SourceType28181 {
-		log.Sugar.Infof("%s 源不是国标推流类型", v.Source)
 		return
 	}

-	var setup gb28181.SetupType
-	switch strings.ToLower(v.Setup) {
-	case "active":
-		setup = gb28181.SetupActive
-		break
-	case "passive":
-		setup = gb28181.SetupPassive
-		break
-	default:
-		setup = gb28181.SetupUDP
-		break
-	}
-
 	addr, _ := net.ResolveTCPAddr("tcp", r.RemoteAddr)
 	sinkId := stream.NetAddr2SinkId(addr)

-	// 添加随机数
+	// sinkId添加随机数
 	if ipv4, ok := sinkId.(uint64); ok {
 		random := uint64(utils.RandomIntInRange(0x1000, 0xFFFF0000))
 		sinkId = (ipv4 & 0xFFFFFFFF00000000) | (random << 16) | (ipv4 & 0xFFFF)
 	}

-	sink, port, err := gb28181.NewForwardSink(v.SSRC, v.Addr, setup, sinkId, v.Source)
+	setup := gb28181.SetupTypeFromString(v.Setup)
+	if v.AnswerSetup != "" {
+		setup = gb28181.SetupTypeFromString(v.AnswerSetup)
+	}
+
+	var protocol stream.TransStreamProtocol
+	// 级联转发
+	if v.SessionName == "" || v.SessionName == InviteTypePlay ||
+		v.SessionName == InviteTypePlayback ||
+		v.SessionName == InviteTypeDownload {
+		protocol = stream.TransStreamGBCascadedForward
+	} else {
+		// 对讲广播转发
+		protocol = stream.TransStreamGBTalkForward
+	}
+
+	var port int
+	sink, port, err = stream.NewForwardSink(setup.TransportType(), protocol, sinkId, v.Source, v.Addr, gb28181.TransportManger)
 	if err != nil {
 		return
 	}

-	source.AddSink(sink)
-
-	log.Sugar.Infof("设置国标级联转发成功 ID: %s", sink.GetID())
+	log.Sugar.Infof("创建转发sink成功, sink: %s port: %d transport: %s", sink.GetID(), port, setup.TransportType())
+	_, state := stream.PreparePlaySink(sink)
+	if utils.HookStateOK != state {
+		err = fmt.Errorf("failed to prepare play sink")
+		return
+	}

 	response := struct {
 		Sink string `json:"sink"` //sink id
-		IP   string `json:"ip"`
-		Port int    `json:"port"`
-	}{Sink: stream.SinkId2String(sinkId), IP: stream.AppConfig.PublicIP, Port: port}
+		SDP
+	}{Sink: stream.SinkId2String(sinkId), SDP: SDP{Addr: net.JoinHostPort(stream.AppConfig.PublicIP, strconv.Itoa(port))}}

 	httpResponseOK(w, &response)
 }
+
+// OnGBTalk 国标广播/对讲流程:
+// 1. 浏览器使用WS携带source_id访问/api/v1/gb28181/talk, 如果source_id冲突, 直接断开ws连接
+// 2. WS链接建立后, 调用gb-cms接口/api/v1/broadcast/invite, 向设备发送广播请求
+func (api *ApiServer) OnGBTalk(w http.ResponseWriter, r *http.Request) {
+	conn, err := api.upgrader.Upgrade(w, r, nil)
+	if err != nil {
+		log.Sugar.Errorf("升级为websocket失败 err: %s", err.Error())
+		conn.Close()
+		return
+	}
+
+	// 获取id
+	id := r.FormValue("source")
+
+	talkSource := gb28181.NewTalkSource(id, conn)
+	talkSource.Init(stream.TCPReceiveBufferQueueSize)
+	talkSource.SetUrlValues(r.Form)
+
+	_, state := stream.PreparePublishSource(talkSource, true)
+	if utils.HookStateOK != state {
+		log.Sugar.Errorf("对讲失败, source: %s", talkSource)
+		conn.Close()
+		return
+	}
+
+	log.Sugar.Infof("ws对讲连接成功, source: %s", talkSource)
+
+	go stream.LoopEvent(talkSource)
+
+	for {
+		_, bytes, err := conn.ReadMessage()
+		length := len(bytes)
+		if err != nil {
+			log.Sugar.Errorf("读取对讲音频包失败, source: %s err: %s", id, err.Error())
+			break
+		} else if length < 1 {
+			continue
+		}
+
+		for i := 0; i < length; {
+			data := stream.UDPReceiveBufferPool.Get().([]byte)
+			n := bufio.MinInt(stream.UDPReceiveBufferSize, length-i)
+			copy(data, bytes[:n])
+			_ = talkSource.PublishSource.Input(data[:n])
+			i += n
+		}
+	}
+
+	talkSource.Close()
+}
--- a/gb28181/cascaded_stream.go
+++ b/gb28181/cascaded_stream.go
@@ -0,0 +1,9 @@
+package gb28181
+
+import (
+	"github.com/lkmio/lkm/stream"
+)
+
+func CascadedTransStreamFactory(source stream.Source, protocol stream.TransStreamProtocol, tracks []*stream.Track) (stream.TransStream, error) {
+	return stream.NewRtpTransStream(stream.TransStreamGBCascadedForward, 1024), nil
+}
--- a/gb28181/forward_sink.go
+++ b/gb28181/forward_sink.go
@@ -1,124 +0,0 @@
-package gb28181
-
-import (
-	"github.com/lkmio/avformat/collections"
-	"github.com/lkmio/avformat/utils"
-	"github.com/lkmio/lkm/log"
-	"github.com/lkmio/lkm/stream"
-	"github.com/lkmio/rtp"
-	"github.com/lkmio/transport"
-	"net"
-)
-
-type ForwardSink struct {
-	stream.BaseSink
-	setup  SetupType
-	socket transport.Transport
-	ssrc   uint32
-}
-
-func (f *ForwardSink) OnConnected(conn net.Conn) []byte {
-	log.Sugar.Infof("级联连接 conn: %s", conn.RemoteAddr())
-
-	f.Conn = conn
-	f.Conn.(*transport.Conn).EnableAsyncWriteMode(512)
-	return nil
-}
-
-func (f *ForwardSink) OnPacket(conn net.Conn, data []byte) []byte {
-	return nil
-}
-
-func (f *ForwardSink) OnDisConnected(conn net.Conn, err error) {
-	log.Sugar.Infof("级联断开连接 conn: %s", conn.RemoteAddr())
-
-	f.Close()
-}
-
-func (f *ForwardSink) Write(index int, data []*collections.ReferenceCounter[[]byte], ts int64, keyVideo bool) error {
-	// TCP等待连接后再转发数据
-	if SetupUDP != f.setup && f.Conn == nil {
-		return nil
-	}
-
-	// 修改为与上级协商的SSRC
-	rtp.ModifySSRC(data[0].Get(), f.ssrc)
-
-	if SetupUDP == f.setup {
-		f.socket.(*transport.UDPClient).Write(data[0].Get()[2:])
-	} else {
-		return f.BaseSink.Write(index, data, ts, keyVideo)
-	}
-
-	return nil
-}
-
-// Close 关闭国标转发流
-func (f *ForwardSink) Close() {
-	f.BaseSink.Close()
-
-	if f.socket != nil {
-		f.socket.Close()
-	}
-}
-
-// NewForwardSink 创建国标级联转发流Sink
-// 返回监听的端口和Sink
-func NewForwardSink(ssrc uint32, serverAddr string, setup SetupType, sinkId stream.SinkID, sourceId string) (stream.Sink, int, error) {
-	sink := &ForwardSink{BaseSink: stream.BaseSink{ID: sinkId, SourceID: sourceId, State: stream.SessionStateCreated, Protocol: stream.TransStreamGBStreamForward}, ssrc: ssrc, setup: setup}
-
-	if SetupUDP == setup {
-		remoteAddr, err := net.ResolveUDPAddr("udp", serverAddr)
-		if err != nil {
-			return nil, 0, err
-		}
-
-		client, err := TransportManger.NewUDPClient(remoteAddr)
-		if err != nil {
-			return nil, 0, err
-		}
-
-		sink.socket = client
-	} else if SetupActive == setup {
-		server, err := TransportManger.NewTCPServer()
-		if err != nil {
-			return nil, 0, err
-		}
-
-		sink.TCPStreaming = true
-		sink.socket = server
-	} else if SetupPassive == setup {
-		client := transport.TCPClient{}
-		err := TransportManger.AllocPort(true, func(port uint16) error {
-			localAddr, err := net.ResolveTCPAddr("tcp", stream.ListenAddr(int(port)))
-			if err != nil {
-				return err
-			}
-
-			remoteAddr, err := net.ResolveTCPAddr("tcp", serverAddr)
-			if err != nil {
-				return err
-			}
-
-			client.SetHandler(sink)
-			conn, err := client.Connect(localAddr, remoteAddr)
-			if err != nil {
-				return err
-			}
-
-			sink.Conn = conn
-			return nil
-		})
-
-		if err != nil {
-			return nil, 0, err
-		}
-
-		sink.TCPStreaming = true
-		sink.socket = &client
-	} else {
-		utils.Assert(false)
-	}
-
-	return sink, sink.socket.ListenPort(), nil
-}
--- a/gb28181/forward_stream.go
+++ b/gb28181/forward_stream.go
@@ -1,61 +0,0 @@
-package gb28181
-
-import (
-	"encoding/binary"
-	"github.com/lkmio/avformat"
-	"github.com/lkmio/avformat/collections"
-	"github.com/lkmio/lkm/log"
-	"github.com/lkmio/lkm/stream"
-)
-
-// ForwardStream 国标级联转发流, 下级推什么, 就向上级发什么.
-type ForwardStream struct {
-	stream.BaseTransStream
-	rtpBuffers *collections.Queue[*collections.ReferenceCounter[[]byte]]
-}
-
-func (f *ForwardStream) WriteHeader() error {
-	return nil
-}
-
-func (f *ForwardStream) Input(packet *avformat.AVPacket) ([]*collections.ReferenceCounter[[]byte], int64, bool, error) {
-	size := 2 + uint16(len(packet.Data))
-	if size > stream.UDPReceiveBufferSize {
-		log.Sugar.Errorf("国标级联转发流失败 rtp包过长, 长度：%d, 最大允许：%d", len(packet.Data), stream.UDPReceiveBufferSize)
-		return nil, 0, false, nil
-	}
-
-	// 释放rtp包
-	for f.rtpBuffers.Size() > 0 {
-		rtp := f.rtpBuffers.Peek(0)
-		if rtp.UseCount() > 1 {
-			break
-		}
-
-		f.rtpBuffers.Pop()
-
-		// 放回池中
-		data := rtp.Get()
-		stream.UDPReceiveBufferPool.Put(data[:cap(data)])
-	}
-
-	bytes := stream.UDPReceiveBufferPool.Get().([]byte)
-	binary.BigEndian.PutUint16(bytes, size-2)
-	copy(bytes[2:], packet.Data)
-
-	rtp := collections.NewReferenceCounter(bytes[:size])
-	f.rtpBuffers.Push(rtp)
-	// 每帧都当关键帧, 直接发给上级
-	return []*collections.ReferenceCounter[[]byte]{rtp}, -1, true, nil
-}
-
-func NewTransStream() (stream.TransStream, error) {
-	return &ForwardStream{
-		BaseTransStream: stream.BaseTransStream{Protocol: stream.TransStreamGBStreamForward},
-		rtpBuffers:      collections.NewQueue[*collections.ReferenceCounter[[]byte]](1024),
-	}, nil
-}
-
-func TransStreamFactory(source stream.Source, protocol stream.TransStreamProtocol, tracks []*stream.Track) (stream.TransStream, error) {
-	return NewTransStream()
-}
--- a/gb28181/publish_test.go
+++ b/gb28181/publish_test.go
@@ -53,13 +53,15 @@ func connectSource(source string, addr string) {

 func createSource(source, setup string, ssrc uint32) (string, uint16, uint32) {
 	v := struct {
-		Source string `json:"source"` //GetSourceID
-		Setup  string `json:"setup"`  //active/passive
-		SSRC   uint32 `json:"ssrc,omitempty"`
+		Source      string `json:"source"` //GetSourceID
+		Setup       string `json:"setup"`  //active/passive
+		SSRC        string `json:"ssrc,omitempty"`
+		SessionName string `json:"session_name,omitempty"` // play/download/playback/talk/broadcast
 	}{
-		Source: source,
-		Setup:  setup,
-		SSRC:   ssrc,
+		Source:      source,
+		Setup:       setup,
+		SSRC:        strconv.Itoa(int(ssrc)),
+		SessionName: "play",
 	}

 	marshal, err := json.Marshal(v)
@@ -67,7 +69,8 @@ func createSource(source, setup string, ssrc uint32) (string, uint16, uint32) {
 		panic(err)
 	}

-	request, err := http.NewRequest("POST", "http://localhost:8080/api/v1/gb28181/source/create", bytes.NewBuffer(marshal))
+	//request, err := http.NewRequest("POST", "http://localhost:8080/api/v1/gb28181/source/create", bytes.NewBuffer(marshal))
+	request, err := http.NewRequest("POST", "http://localhost:8080/api/v1/gb28181/offer/create", bytes.NewBuffer(marshal))
 	if err != nil {
 		panic(err)
 	}
@@ -89,8 +92,7 @@ func createSource(source, setup string, ssrc uint32) (string, uint16, uint32) {
 		Code int    `json:"code"`
 		Msg  string `json:"msg"`
 		Data struct {
-			IP   string `json:"ip"`
-			Port uint16 `json:"port,omitempty"`
+			Addr string `json:"addr"`
 			SSRC string `json:"ssrc,omitempty"`
 		}
 	}{}
@@ -105,7 +107,9 @@ func createSource(source, setup string, ssrc uint32) (string, uint16, uint32) {
 		panic(err)
 	}

-	return connectInfo.Data.IP, connectInfo.Data.Port, uint32(atoi)
+	host, p, err := net.SplitHostPort(connectInfo.Data.Addr)
+	Port, err := strconv.Atoi(p)
+	return host, uint16(Port), uint32(atoi)
 }

 // 分割rtp包, 返回rtp over tcp包
--- a/gb28181/source.go
+++ b/gb28181/source.go
@@ -25,6 +25,43 @@ const (
 	JitterBufferSize  = 1024 * 1024
 )

+func (s SetupType) TransportType() stream.TransportType {
+	switch s {
+	case SetupUDP:
+		return stream.TransportTypeUDP
+	case SetupPassive:
+		return stream.TransportTypeTCPServer
+	case SetupActive:
+		return stream.TransportTypeTCPClient
+	default:
+		panic(fmt.Errorf("invalid setup type: %d", s))
+	}
+}
+
+func (s SetupType) String() string {
+	switch s {
+	case SetupUDP:
+		return "udp"
+	case SetupPassive:
+		return "passive"
+	case SetupActive:
+		return "active"
+	default:
+		panic(fmt.Errorf("invalid setup type: %d", s))
+	}
+}
+
+func SetupTypeFromString(setupType string) SetupType {
+	switch setupType {
+	case "passive":
+		return SetupPassive
+	case "active":
+		return SetupActive
+	default:
+		return SetupUDP
+	}
+}
+
 var (
 	TransportManger transport.Manager
 	SharedUDPServer *UDPServer
--- a/gb28181/talk_source.go
+++ b/gb28181/talk_source.go
@@ -0,0 +1,94 @@
+package gb28181
+
+import (
+	"github.com/gorilla/websocket"
+	"github.com/lkmio/avformat"
+	"github.com/lkmio/avformat/bufio"
+	"github.com/lkmio/avformat/utils"
+	"github.com/lkmio/lkm/stream"
+	"time"
+)
+
+type Demuxer struct {
+	avformat.BaseDemuxer
+	ts            int64
+	firstOfPacket bool
+}
+
+func (d *Demuxer) Input(data []byte) (int, error) {
+	length := len(data)
+
+	if !d.firstOfPacket {
+		d.firstOfPacket = true
+		d.OnNewAudioTrack(0, utils.AVCodecIdPCMALAW, 8000, nil, avformat.AudioConfig{
+			HasADTSHeader: false,
+			Channels:      1,
+			SampleRate:    8000,
+			SampleSize:    2,
+		})
+
+		d.ProbeComplete()
+	}
+
+	for i := 0; i < length; {
+		n := bufio.MinInt(length-i, 320)
+		_, _ = d.DataPipeline.Write(data[i:i+n], 0, utils.AVMediaTypeAudio)
+		pkt, _ := d.DataPipeline.Feat(0)
+		d.OnAudioPacket(0, utils.AVCodecIdPCMALAW, pkt, d.ts)
+		d.ts += int64(n)
+		i += n
+	}
+
+	return length, nil
+}
+
+type TalkSource struct {
+	stream.PublishSource
+}
+
+func (s *TalkSource) Input(data []byte) error {
+	_, err := s.PublishSource.TransDemuxer.Input(data)
+	return err
+}
+
+func (s *TalkSource) Close() {
+	s.PublishSource.Close()
+	// 关闭所有对讲设备的会话
+	stream.CloseWaitingSinks(s.ID)
+}
+
+type WSConn struct {
+	*websocket.Conn
+}
+
+func (w WSConn) Read(b []byte) (n int, err error) {
+	panic("implement me")
+}
+
+func (w WSConn) Write(block []byte) (n int, err error) {
+	panic("implement me")
+}
+
+func (w WSConn) SetDeadline(t time.Time) error {
+	panic("implement me")
+}
+
+func NewTalkSource(id string, conn *websocket.Conn) *TalkSource {
+	s := &TalkSource{
+		PublishSource: stream.PublishSource{
+			ID:   id,
+			Type: stream.SourceTypeGBTalk,
+			Conn: &WSConn{conn},
+			TransDemuxer: &Demuxer{
+				BaseDemuxer: avformat.BaseDemuxer{
+					DataPipeline: &avformat.StreamsBuffer{},
+					Name:         "gb_talk",
+					AutoFree:     false,
+				},
+			},
+		},
+	}
+
+	s.TransDemuxer.SetHandler(s)
+	return s
+}
--- a/gb28181/talk_stream.go
+++ b/gb28181/talk_stream.go
@@ -0,0 +1,42 @@
+package gb28181
+
+import (
+	"github.com/lkmio/avformat"
+	"github.com/lkmio/avformat/collections"
+	"github.com/lkmio/lkm/stream"
+	"github.com/lkmio/rtp"
+)
+
+type TalkStream struct {
+	*stream.RtpStream
+	muxer  rtp.Muxer
+	packet []byte
+}
+
+func (s *TalkStream) WriteHeader() error {
+	return nil
+}
+
+func (s *TalkStream) Input(packet *avformat.AVPacket) ([]*collections.ReferenceCounter[[]byte], int64, bool, error) {
+	var size int
+	s.muxer.Input(packet.Data, uint32(packet.Dts), func() []byte {
+		return s.packet
+	}, func(pkt []byte) {
+		size = len(pkt)
+	})
+
+	packet = &avformat.AVPacket{Data: s.packet[:size]}
+	return s.RtpStream.Input(packet)
+}
+
+func NewTalkTransStream() (stream.TransStream, error) {
+	return &TalkStream{
+		RtpStream: stream.NewRtpTransStream(stream.TransStreamGBTalkForward, 1024),
+		muxer:     rtp.NewMuxer(8, 0, 0xFFFFFFFF),
+		packet:    make([]byte, 1500),
+	}, nil
+}
+
+func TalkTransStreamFactory(source stream.Source, protocol stream.TransStreamProtocol, tracks []*stream.Track) (stream.TransStream, error) {
+	return NewTalkTransStream()
+}
--- a/main.go
+++ b/main.go
@@ -28,7 +28,8 @@ func init() {
 	stream.RegisterTransStreamFactory(stream.TransStreamFlv, flv.TransStreamFactory)
 	stream.RegisterTransStreamFactory(stream.TransStreamRtsp, rtsp.TransStreamFactory)
 	stream.RegisterTransStreamFactory(stream.TransStreamRtc, rtc.TransStreamFactory)
-	stream.RegisterTransStreamFactory(stream.TransStreamGBStreamForward, gb28181.TransStreamFactory)
+	stream.RegisterTransStreamFactory(stream.TransStreamGBCascadedForward, gb28181.CascadedTransStreamFactory)
+	stream.RegisterTransStreamFactory(stream.TransStreamGBTalkForward, gb28181.TalkTransStreamFactory)
 	stream.SetRecordStreamFactory(record.NewFLVFileSink)
 	stream.StreamEndInfoBride = NewStreamEndInfo

--- a/stream/forward_sink.go
+++ b/stream/forward_sink.go
@@ -0,0 +1,161 @@
+package stream
+
+import (
+	"github.com/lkmio/avformat/collections"
+	"github.com/lkmio/lkm/log"
+	"github.com/lkmio/transport"
+	"net"
+	"time"
+)
+
+type TransportType int
+
+const (
+	TransportTypeUDP TransportType = iota
+	TransportTypeTCPClient
+	TransportTypeTCPServer
+)
+
+func (t TransportType) String() string {
+	switch t {
+	case TransportTypeUDP:
+		return "udp"
+	case TransportTypeTCPClient:
+		return "tcp_client"
+	case TransportTypeTCPServer:
+		return "tcp_server"
+	default:
+		panic("invalid transport type")
+	}
+}
+
+type ForwardSink struct {
+	BaseSink
+	socket        transport.Transport
+	transportType TransportType
+	receiveTimer  *time.Timer
+}
+
+func (f *ForwardSink) OnConnected(conn net.Conn) []byte {
+	log.Sugar.Infof("%s 连接 conn: %s", f.Protocol, conn.RemoteAddr())
+
+	f.receiveTimer.Stop()
+
+	// 如果f.Conn赋值后, 发送数据先于EnableAsyncWriteMode执行, 可能会panic
+	// 所以保险一点, 放在主协程执行
+	ExecuteSyncEventOnSource(f.SourceID, func() {
+		f.Conn = conn
+		f.BaseSink.EnableAsyncWriteMode(512)
+	})
+
+	return nil
+}
+
+func (f *ForwardSink) OnPacket(conn net.Conn, data []byte) []byte {
+	return nil
+}
+
+func (f *ForwardSink) OnDisConnected(conn net.Conn, err error) {
+	log.Sugar.Infof("%s 断开连接 conn: %s", f.Protocol, conn.RemoteAddr())
+
+	f.Close()
+}
+
+func (f *ForwardSink) Write(index int, data []*collections.ReferenceCounter[[]byte], ts int64, keyVideo bool) error {
+	// TCP等待连接后再转发数据
+	if TransportTypeUDP != f.transportType && f.Conn == nil {
+		return nil
+	}
+
+	if TransportTypeUDP == f.transportType {
+		f.socket.(*transport.UDPClient).Write(data[0].Get()[2:])
+	} else {
+		return f.BaseSink.Write(index, data, ts, keyVideo)
+	}
+
+	return nil
+}
+
+// Close 关闭国标转发流
+func (f *ForwardSink) Close() {
+	f.BaseSink.Close()
+
+	if f.socket != nil {
+		f.socket.Close()
+	}
+
+	if f.receiveTimer != nil {
+		f.receiveTimer.Stop()
+	}
+}
+
+// StartReceiveTimer 启动tcp sever计时器, 如果计时器触发, 没有连接, 则关闭流
+func (f *ForwardSink) StartReceiveTimer() {
+	f.receiveTimer = time.AfterFunc(time.Second*10, func() {
+		if f.Conn == nil {
+			log.Sugar.Infof("%s 等待连接超时, 关闭sink", f.Protocol)
+			f.Close()
+		}
+	})
+}
+
+func NewForwardSink(transportType TransportType, protocol TransStreamProtocol, sinkId SinkID, sourceId string, addr string, manager transport.Manager) (*ForwardSink, int, error) {
+	sink := &ForwardSink{
+		BaseSink:      BaseSink{ID: sinkId, SourceID: sourceId, State: SessionStateCreated, Protocol: protocol},
+		transportType: transportType,
+	}
+
+	if transportType == TransportTypeUDP {
+		remoteAddr, err := net.ResolveUDPAddr("udp", addr)
+		if err != nil {
+			return nil, 0, err
+		}
+
+		client, err := manager.NewUDPClient(remoteAddr)
+		if err != nil {
+			return nil, 0, err
+		}
+
+		sink.socket = client
+	} else if transportType == TransportTypeTCPClient {
+		client := transport.TCPClient{}
+		err := manager.AllocPort(true, func(port uint16) error {
+			localAddr, err := net.ResolveTCPAddr("tcp", ListenAddr(int(port)))
+			if err != nil {
+				return err
+			}
+
+			remoteAddr, err := net.ResolveTCPAddr("tcp", addr)
+			if err != nil {
+				return err
+			}
+
+			client.SetHandler(sink)
+			conn, err := client.Connect(localAddr, remoteAddr)
+			if err != nil {
+				return err
+			}
+
+			sink.Conn = conn
+			return nil
+		})
+
+		if err != nil {
+			return nil, 0, err
+		}
+
+		sink.socket = &client
+	} else if transportType == TransportTypeTCPServer {
+		tcpServer, err := manager.NewTCPServer()
+		if err != nil {
+			return nil, 0, err
+		}
+
+		tcpServer.SetHandler(sink)
+		tcpServer.Accept()
+		sink.socket = tcpServer
+		sink.StartReceiveTimer()
+	}
+
+	return sink, sink.socket.ListenPort(), nil
+}
--- a/stream/rtp_stream.go
+++ b/stream/rtp_stream.go
@@ -0,0 +1,56 @@
+package stream
+
+import (
+	"encoding/binary"
+	"github.com/lkmio/avformat"
+	"github.com/lkmio/avformat/collections"
+	"github.com/lkmio/lkm/log"
+)
+
+type RtpStream struct {
+	BaseTransStream
+	rtpBuffers *collections.Queue[*collections.ReferenceCounter[[]byte]]
+}
+
+func (f *RtpStream) WriteHeader() error {
+	return nil
+}
+
+func (f *RtpStream) Input(packet *avformat.AVPacket) ([]*collections.ReferenceCounter[[]byte], int64, bool, error) {
+	size := 2 + uint16(len(packet.Data))
+	if size > UDPReceiveBufferSize {
+		log.Sugar.Errorf("转发%s流失败 rtp包过长, 长度：%d, 最大允许：%d", f.Protocol, len(packet.Data), UDPReceiveBufferSize)
+		return nil, 0, false, nil
+	}
+
+	// 释放rtp包
+	for f.rtpBuffers.Size() > 0 {
+		rtp := f.rtpBuffers.Peek(0)
+		if rtp.UseCount() > 1 {
+			break
+		}
+
+		f.rtpBuffers.Pop()
+
+		// 放回池中
+		data := rtp.Get()
+		UDPReceiveBufferPool.Put(data[:cap(data)])
+	}
+
+	bytes := UDPReceiveBufferPool.Get().([]byte)
+	binary.BigEndian.PutUint16(bytes, size-2)
+	copy(bytes[2:], packet.Data)
+
+	rtp := collections.NewReferenceCounter(bytes[:size])
+	f.rtpBuffers.Push(rtp)
+
+	// 每帧都当关键帧, 直接发给上级
+	return []*collections.ReferenceCounter[[]byte]{rtp}, -1, true, nil
+}
+
+func NewRtpTransStream(protocol TransStreamProtocol, capacity int) *RtpStream {
+	return &RtpStream{
+		BaseTransStream: BaseTransStream{Protocol: protocol},
+		rtpBuffers:      collections.NewQueue[*collections.ReferenceCounter[[]byte]](capacity),
+	}
+}
--- a/stream/sink_utils.go
+++ b/stream/sink_utils.go
@@ -53,3 +53,13 @@ func SinkId2String(id SinkID) string {
 func CreateSinkDisconnectionMessage(sink Sink) string {
 	return fmt.Sprintf("%s sink断开连接. id: %s", sink.GetProtocol(), sink.GetID())
 }
+
+func ExecuteSyncEventOnSource(sourceId string, event func()) bool {
+	source := SourceManager.Find(sourceId)
+	if source != nil {
+		source.ExecuteSyncEvent(event)
+		return true
+	}
+
+	return false
+}
--- a/stream/sink_waitting.go
+++ b/stream/sink_waitting.go
@@ -63,6 +63,13 @@ func PopWaitingSinks(sourceId string) []Sink {
 	return sinks
 }

+func CloseWaitingSinks(sourceId string) {
+	sinks := PopWaitingSinks(sourceId)
+	for _, sink := range sinks {
+		sink.Close()
+	}
+}
+
 func ExistSinkInWaitingQueue(sourceId string, sinkId SinkID) bool {
 	mutex.RLock()
 	defer mutex.RUnlock()
--- a/stream/source.go
+++ b/stream/source.go
@@ -288,7 +288,7 @@ func (s *PublishSource) CreateTransStream(id TransStreamID, protocol TransStream
 	_ = transStream.WriteHeader()

 	// 设置转发流
-	if TransStreamGBStreamForward == transStream.GetProtocol() {
+	if TransStreamGBCascadedForward == transStream.GetProtocol() {
 		s.ForwardTransStream = transStream
 	}

@@ -612,7 +612,7 @@ func (s *PublishSource) DoClose() {
 		transStreamID := sink.GetTransStreamID()
 		sink.SetTransStreamID(0)
 		if s.recordSink == sink {
-			return
+			continue
 		}

 		{
--- a/stream/source_utils.go
+++ b/stream/source_utils.go
@@ -20,16 +20,18 @@ type TransStreamProtocol uint32
 type SessionState uint32

 const (
-	SourceTypeRtmp  = SourceType(1)
-	SourceType28181 = SourceType(2)
-	SourceType1078  = SourceType(3)
+	SourceTypeRtmp   = SourceType(1)
+	SourceType28181  = SourceType(2)
+	SourceType1078   = SourceType(3)
+	SourceTypeGBTalk = SourceType(4) // 国标广播/对讲

-	TransStreamRtmp            = TransStreamProtocol(1)
-	TransStreamFlv             = TransStreamProtocol(2)
-	TransStreamRtsp            = TransStreamProtocol(3)
-	TransStreamHls             = TransStreamProtocol(4)
-	TransStreamRtc             = TransStreamProtocol(5)
-	TransStreamGBStreamForward = TransStreamProtocol(6) // 国标级联转发
+	TransStreamRtmp              = TransStreamProtocol(1)
+	TransStreamFlv               = TransStreamProtocol(2)
+	TransStreamRtsp              = TransStreamProtocol(3)
+	TransStreamHls               = TransStreamProtocol(4)
+	TransStreamRtc               = TransStreamProtocol(5)
+	TransStreamGBCascadedForward = TransStreamProtocol(6) // 国标级联转发
+	TransStreamGBTalkForward     = TransStreamProtocol(7) // 国标广播/对讲转发
 )

 const (
@@ -49,6 +51,8 @@ func (s SourceType) String() string {
 		return "28181"
 	} else if SourceType1078 == s {
 		return "jt1078"
+	} else if SourceTypeGBTalk == s {
+		return "gb_talk"
 	}

 	panic(fmt.Sprintf("unknown source type %d", s))
@@ -65,8 +69,10 @@ func (p TransStreamProtocol) String() string {
 		return "hls"
 	} else if TransStreamRtc == p {
 		return "rtc"
-	} else if TransStreamGBStreamForward == p {
-		return "gb_stream_forward"
+	} else if TransStreamGBCascadedForward == p {
+		return "gb_cascaded_forward"
+	} else if TransStreamGBTalkForward == p {
+		return "gb_talk_forward"
 	}

 	panic(fmt.Sprintf("unknown stream protocol %d", p))
--- a/web/broadcast.html
+++ b/web/broadcast.html
@@ -0,0 +1,278 @@
+<!DOCTYPE html>
+<html>
+<head>
+    <meta charset="UTF-8"/>
+    <meta name="viewport"
+          content="width=device-width, user-scalable=no, initial-scale=1.0, maximum-scale=1.0, minimum-scale=1.0"/>
+    <meta name="apple-mobile-web-capable" content="yes"/>
+    <title>语音广播</title>
+</head>
+<body>
+<button id="intercomBegin">开始广播</button>
+<button id="intercomEnd">关闭广播</button>
+
+<input style="width: 100px;" id="device_id" type="text" value="34020000001320000001"/>
+<input style="width: 100px;" id="channel_id" type="text" value="34020000001320000001"/>
+
+<button id="invite" onclick="invite()">邀请</button>
+<button id="hangup" onclick="hangup()">挂断</button>
+
+</body>
+<script src="g711.js"></script>
+<script type="text/javascript">
+
+    var begin = document.getElementById('intercomBegin');
+    var end = document.getElementById('intercomEnd');
+
+    var ws = null; //实现WebSocket
+    var record = null; //多媒体对象，用来处理音频
+    var source = null;
+
+    function init(rec) {
+        record = rec;
+    }
+
+    function invite() {
+        let deviceId = document.getElementById("device_id").value;
+        let channelId = document.getElementById("channel_id").value;
+
+        let data = {
+            device_id: deviceId,
+            channel_id: channelId,
+            type: 1,
+            source: source
+        };
+
+        fetch("http://localhost:9000/api/v1/broadcast/invite", {
+            method: 'POST',
+            body: JSON.stringify(data),
+            headers: new Headers({
+                'Content-Type': 'application/json',
+            }),
+        }).then((res) => res.json())
+            .then((data) => {
+
+            })
+    }
+
+    function hangup() {
+        let deviceId = document.getElementById("device_id").value;
+        let channelId = document.getElementById("channel_id").value;
+
+        let data = {
+            device_id: deviceId,
+            channel_id: channelId,
+            source: source
+        };
+
+        fetch("http://localhost:9000/api/v1/broadcast/hangup", {
+            method: 'POST',
+            body: JSON.stringify(data),
+            headers: new Headers({
+                'Content-Type': 'application/json',
+            }),
+        }).then((res) => res.json())
+            .then((data) => {
+
+            })
+    }
+
+    //录音对象
+    var Recorder = function (stream) {
+        var sampleBits = 16; //输出采样数位 8, 16
+        var sampleRate = 8000; //输出采样率
+        var context = new AudioContext();
+        var audioInput = context.createMediaStreamSource(stream);
+        var recorder = context.createScriptProcessor(4096, 1, 1);
+        var audioData = {
+            size: 0, //录音文件长度
+            buffer: [], //录音缓存
+            inputSampleRate: 48000, //输入采样率
+            inputSampleBits: 16, //输入采样数位 8, 16
+            outputSampleRate: sampleRate, //输出采样数位
+            oututSampleBits: sampleBits, //输出采样率
+            clear: function () {
+                this.buffer = [];
+                this.size = 0;
+            },
+            input: function (data) {
+                this.buffer.push(new Float32Array(data));
+                this.size += data.length;
+            },
+            compress: function () { //合并压缩
+                //合并
+                var data = new Float32Array(this.size);
+                var offset = 0;
+                for (var i = 0; i < this.buffer.length; i++) {
+                    data.set(this.buffer[i], offset);
+                    offset += this.buffer[i].length;
+                }
+                //压缩
+                var compression = parseInt(this.inputSampleRate / this.outputSampleRate);
+                var length = data.length / compression;
+                var result = new Float32Array(length);
+                var index = 0,
+                    j = 0;
+                while (index < length) {
+                    result[index] = data[j];
+                    j += compression;
+                    index++;
+                }
+                return result;
+            },
+            encodePCM: function () { //这里不对采集到的数据进行其他格式处理，如有需要均交给服务器端处理。
+                var sampleRate = Math.min(this.inputSampleRate, this.outputSampleRate);
+                var sampleBits = Math.min(this.inputSampleBits, this.oututSampleBits);
+                var bytes = this.compress();
+                var dataLength = bytes.length * (sampleBits / 8);
+                var buffer = new ArrayBuffer(dataLength);
+                var data = new DataView(buffer);
+                var offset = 0;
+                for (var i = 0; i < bytes.length; i++, offset += 2) {
+                    var s = Math.max(-1, Math.min(1, bytes[i]));
+                    data.setInt16(offset, s < 0 ? s * 0x8000 : s * 0x7FFF, true);
+                }
+                return new Blob([data]);
+            }
+        };
+
+        var sendData = function () { //对以获取的数据进行处理(分包)
+            var reader = new FileReader();
+            reader.onload = e => {
+                var outbuffer = e.target.result;
+                var arr = new Uint16Array(outbuffer);
+                var dst = new Int8Array(arr.length);
+                pcm16_to_alaw(arr.byteLength, arr, dst);
+
+                if (ws && ws.readyState === WebSocket.OPEN) {
+                    ws.send(dst);
+                } else {
+                    console.log('WebSocket not ready, state:', ws ? ws.readyState : 'null');
+                    record.stop(); // 停止录音
+                    if (ws) {
+                        ws.close(); // 确保关闭连接
+                    }
+                }
+            };
+
+            reader.readAsArrayBuffer(audioData.encodePCM());
+            audioData.clear();//每次发送完成则清理掉旧数据
+        };
+
+        this.start = function () {
+            audioInput.connect(recorder);
+            recorder.connect(context.destination);
+        }
+
+        this.stop = function () {
+            recorder.disconnect();
+        }
+
+        this.getBlob = function () {
+            return audioData.encodePCM();
+        }
+
+        this.clear = function () {
+            audioData.clear();
+        }
+
+        recorder.onaudioprocess = function (e) {
+            var inputBuffer = e.inputBuffer.getChannelData(0);
+            audioData.input(inputBuffer);
+            sendData();
+        }
+    }
+
+    function generateRandomAlphanumeric10() {
+        const chars = 'ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789';
+        let result = '';
+        for (let i = 0; i < 20; i++) {
+            result += chars.charAt(Math.floor(Math.random() * chars.length));
+        }
+        return result;
+    }
+
+    /*
+    * WebSocket
+    */
+    function connectWS() {
+        let secure = window.location.protocol === 'https:'
+        //source = generateRandomAlphanumeric10();
+        source = "unique_id";
+        let url = (secure ? "wss:" : "ws:") + "/" + window.location.host + "/ws/v1/gb28181/talk" + "?source=" + source
+        ws = new WebSocket(url);
+        ws.binaryType = 'arraybuffer'; //传输的是 ArrayBuffer 类型的数据
+
+        ws.onopen = function () {
+            console.log('ws连接成功');
+            record.start();
+        };
+
+        ws.onmessage = function (msg) {
+
+        }
+
+        ws.onerror = function (err) {
+            console.log('ws连接断开');
+            console.info(err)
+            record.stop()
+        }
+    }
+
+    /*
+    * 开始对讲
+    */
+    begin.onclick = function () {
+        navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia;
+        if (!navigator.getUserMedia) {
+            alert('浏览器不支持音频输入');
+        } else {
+            navigator.getUserMedia({
+                    audio: true
+                },
+
+                //获取到音频采集权限回调
+                function (mediaStream) {
+                    console.log('开始对讲');
+                    //初始化采集器
+                    init(new Recorder(mediaStream));
+                    //连接websocket
+                    connectWS();
+                },
+
+                function (error) {
+                    console.log(error);
+                    switch (error.message || error.name) {
+                        case 'PERMISSION_DENIED':
+                        case 'PermissionDeniedError':
+                            console.info('用户拒绝提供信息。');
+                            break;
+                        case 'NOT_SUPPORTED_ERROR':
+                        case 'NotSupportedError':
+                            console.info('浏览器不支持硬件设备。');
+                            break;
+                        case 'MANDATORY_UNSATISFIED_ERROR':
+                        case 'MandatoryUnsatisfiedError':
+                            console.info('无法发现指定的硬件设备。');
+                            break;
+                        default:
+                            console.info('无法打开麦克风。异常信息:' + (error.code || error.name));
+                            break;
+                    }
+                }
+            )
+        }
+    }
+
+    /*
+    * 关闭对讲
+    */
+    end.onclick = function () {
+        if (ws) {
+            ws.close();
+            record.stop();
+            console.log('关闭对讲以及WebSocket');
+        }
+    }
+</script>
+</html>
--- a/web/g711.js
+++ b/web/g711.js
@@ -0,0 +1,341 @@
+var SIGN_BIT	= 0x80;		/* Sign bit for a A-law byte. */
+var QUANT_MASK	= 0xf;		/* Quantization field mask. */
+var NSEGS		= 0x8;		/* Number of A-law segments. */
+var SEG_SHIFT	= 0x4;		/* Left shift for segment number. */
+var SEG_MASK	= 0x70;		/* Segment field mask. */
+
+
+var seg_aend = new Int16Array([0x1F, 0x3F, 0x7F, 0xFF,0x1FF, 0x3FF, 0x7FF, 0xFFF]);
+
+var seg_uend = new Int16Array([0x3F, 0x7F, 0xFF, 0x1FF,0x3FF, 0x7FF, 0xFFF, 0x1FFF]);
+
+
+ /* copy from CCITT G.711 specifications */
+var _u2a = new Uint8Array([1,	1,	2,	2,	3,	3,	4,	4,
+5,	5,	6,	6,	7,	7,	8,	8,
+9,	10,	11,	12,	13,	14,	15,	16,
+17,	18,	19,	20,	21,	22,	23,	24,
+25,	27,	29,	31,	33,	34,	35,	36,
+37,	38,	39,	40,	41,	42,	43,	44,
+46,	48,	49,	50,	51,	52,	53,	54,
+55,	56,	57,	58,	59,	60,	61,	62,
+64,	65,	66,	67,	68,	69,	70,	71,
+72,	73,	74,	75,	76,	77,	78,	79,
+80,	82,	83,	84,	85,	86,	87,	88,
+89,	90,	91,	92,	93,	94,	95,	96,
+97,	98,	99,	100,	101,	102,	103,	104,
+105,	106,	107,	108,	109,	110,	111,	112,
+113,	114,	115,	116,	117,	118,	119,	120,
+121,	122,	123,	124,	125,	126,	127,	128]);
+
+/* A- to u-law conversions */
+var _a2u = new Uint8Array([1,	3,	5,	7,	9,	11,	13,	15,
+16,	17,	18,	19,	20,	21,	22,	23,
+24,	25,	26,	27,	28,	29,	30,	31,
+32,	32,	33,	33,	34,	34,	35,	35,
+36,	37,	38,	39,	40,	41,	42,	43,
+44,	45,	46,	47,	48,	48,	49,	49,
+50,	51,	52,	53,	54,	55,	56,	57,
+58,	59,	60,	61,	62,	63,	64,	64,
+65,	66,	67,	68,	69,	70,	71,	72,
+73,	74,	75,	76,	77,	78,	79,	80,
+80,	81,	82,	83,	84,	85,	86,	87,
+88,	89,	90,	91,	92,	93,	94,	95,
+96,	97,	98,	99,	100,	101,	102,	103,
+104,	105,	106,	107,	108,	109,	110,	111,
+112,	113,	114,	115,	116,	117,	118,	119,
+120,	121,	122,	123,	124,	125,	126,	127]);
+
+function search( val, table, size)
+{
+    var i;
+    
+    for (i = 0; i < size; i++) {
+        if (val <= table[i])
+        return (i);
+    }
+
+    return (size);
+}
+
+function linear2alaw(pcm_val)	/* 2's complement (16-bit range) */
+{
+var	 mask;
+var	 seg;
+var aval;
+
+pcm_val = pcm_val >> 3;
+
+if (pcm_val >= 0) {
+  mask = 0xD5;		/* sign (7th) bit = 1 */
+} else {
+  mask = 0x55;		/* sign bit = 0 */
+  pcm_val = -pcm_val - 1;
+}
+
+/* Convert the scaled magnitude to segment number. */
+seg = search(pcm_val, seg_aend, 8);
+
+/* Combine the sign, segment, and quantization bits. */
+
+if (seg >= 8)		/* out of range, return maximum value. */
+  return (0x7F ^ mask);
+else {
+  aval = seg << SEG_SHIFT;
+  if (seg < 2)
+ aval |= (pcm_val >> 1) & QUANT_MASK;
+  else
+ aval |= (pcm_val >> seg) & QUANT_MASK;
+  return (aval ^ mask);
+}
+}
+
+/*
+* alaw2linear() - Convert an A-law value to 16-bit linear PCM
+*
+*/
+
+function alaw2linear(a_val){
+var t;
+var seg;
+
+a_val ^= 0x55;
+
+t = (a_val & QUANT_MASK) << 4;
+seg = (a_val & SEG_MASK) >> SEG_SHIFT;
+switch (seg) {
+case 0:
+  t += 8;
+  break;
+case 1:
+  t += 0x108;
+  break;
+default:
+  t += 0x108;
+  t <<= seg - 1;
+}
+return ((a_val & SIGN_BIT) ? t : -t);
+}
+
+
+// #define	BIAS		(0x84)		/* Bias for linear code. */
+// #define CLIP            8159
+var BIAS = 0x84;
+
+var CLIP = 8159;
+
+// /*
+// * linear2ulaw() - Convert a linear PCM value to u-law
+// *
+// * In order to simplify the encoding process, the original linear magnitude
+// * is biased by adding 33 which shifts the encoding range from (0 - 8158) to
+// * (33 - 8191). The result can be seen in the following encoding table:
+// *
+// *	Biased Linear Input Code	Compressed Code
+// *	------------------------	---------------
+// *	00000001wxyza			000wxyz
+// *	0000001wxyzab			001wxyz
+// *	000001wxyzabc			010wxyz
+// *	00001wxyzabcd			011wxyz
+// *	0001wxyzabcde			100wxyz
+// *	001wxyzabcdef			101wxyz
+// *	01wxyzabcdefg			110wxyz
+// *	1wxyzabcdefgh			111wxyz
+// *
+// * Each biased linear code has a leading 1 which identifies the segment
+// * number. The value of the segment number is equal to 7 minus the number
+// * of leading 0's. The quantization interval is directly available as the
+// * four bits wxyz.  * The trailing bits (a - h) are ignored.
+// *
+// * Ordinarily the complement of the resulting code word is used for
+// * transmission, and so the code word is complemented before it is returned.
+// *
+// * For further information see John C. Bellamy's Digital Telephony, 1982,
+// * John Wiley & Sons, pps 98-111 and 472-476.
+function linear2ulaw(pcm_val)	/* 2's complement (16-bit range) */
+{
+var         mask;
+var	 seg;
+var uval;
+
+/* Get the sign and the magnitude of the value. */
+pcm_val = pcm_val >> 2;
+if (pcm_val < 0) {
+  pcm_val = -pcm_val;
+  mask = 0x7F;
+} else {
+  mask = 0xFF;
+}
+if ( pcm_val > CLIP ) pcm_val = CLIP;		/* clip the magnitude */
+pcm_val += (BIAS >> 2);
+
+/* Convert the scaled magnitude to segment number. */
+seg = search(pcm_val, seg_uend, 8);
+
+/*
+* Combine the sign, segment, quantization bits;
+* and complement the code word.
+*/
+if (seg >= 8)		/* out of range, return maximum value. */
+  return (0x7F ^ mask);
+else {
+  uval = (seg << 4) | ((pcm_val >> (seg + 1)) & 0xF);
+  return (uval ^ mask);
+}
+
+}
+
+
+
+// /*
+//  * ulaw2linear() - Convert a u-law value to 16-bit linear PCM
+//  *
+//  * First, a biased linear code is derived from the code word. An unbiased
+//  * output can then be obtained by subtracting 33 from the biased code.
+//  *
+//  * Note that this function expects to be passed the complement of the
+//  * original code word. This is in keeping with ISDN conventions.
+//  */
+function ulaw2linear(u_val)
+{
+var t;
+
+/* Complement to obtain normal u-law value. */
+u_val = ~u_val;
+
+/*
+* Extract and bias the quantization bits. Then
+* shift up by the segment number and subtract out the bias.
+*/
+t = ((u_val & QUANT_MASK) << 3) + BIAS;
+t <<= (u_val & SEG_MASK) >> SEG_SHIFT;
+
+return ((u_val & SIGN_BIT) ? (BIAS - t) : (t - BIAS));
+}
+
+// /* A-law to u-law conversion */
+function alaw2ulaw(aval)
+{
+aval &= 0xff;
+return  ((aval & 0x80) ? (0xFF ^ _a2u[aval ^ 0xD5]) :
+   (0x7F ^ _a2u[aval ^ 0x55]));
+}
+
+/* u-law to A-law conversion */
+function ulaw2alaw(uval)
+{
+uval &= 0xff;
+return  ((uval & 0x80) ? (0xD5 ^ (_u2a[0xFF ^ uval] - 1)) :
+           (0x55 ^ (_u2a[0x7F ^ uval] - 1)));
+}
+
+// unsigned char linear_to_alaw[65536];
+// unsigned char linear_to_ulaw[65536];
+
+var short_index = new Int16Array(65536);
+
+var linear_to_alaw = new Uint8Array(65536);
+var linear_to_ulaw = new Uint8Array(65536);
+
+// /* 16384 entries per table (8 bit) */
+// unsigned short alaw_to_linear[256];
+// unsigned short ulaw_to_linear[256];
+
+var alaw_to_linear =  new Uint8Array(256);
+var ulaw_to_linear =  new Uint8Array(256);
+
+function  build_linear_to_xlaw_table(linear_to_xlaw,linear2xlaw)
+{
+    var i;
+    for (i=0; i<65536;i++){
+        var v = linear2xlaw(short_index[i]);
+        linear_to_xlaw[i] = v;
+    }
+}
+
+
+function build_xlaw_to_linear_table(xlaw_to_linear,xlaw2linear)
+{
+    var i;
+
+    for (i=0; i<256;i++){
+        xlaw_to_linear[i] =  xlaw2linear(i);
+    }
+}
+
+function pcm16_to_xlaw(linear_to_xlaw,  src_length,src_samples,dst_samples)
+{
+    var i;
+    var s_samples;
+
+    s_samples = src_samples;
+
+    for (i=0; i < src_length / 2; i++)
+    {
+        dst_samples[i] = linear_to_xlaw[s_samples[i]];
+    }
+}
+
+function xlaw_to_pcm16(xlaw_to_linear, src_length,src_samples, dst_samples)
+{
+    var i;
+    var  s_samples;
+    var d_samples;
+
+    s_samples =  src_samples;
+    d_samples = dst_samples;
+
+    for (i=0; i < src_length; i++)
+    {
+        d_samples[i] = xlaw_to_linear[s_samples[i]];
+    }
+}
+
+function pcm16_to_alaw(src_length, src_samples, dst_samples)
+{
+    pcm16_to_xlaw(linear_to_alaw, src_length, src_samples, dst_samples);
+}
+
+function pcm16_to_ulaw(src_length, src_samples, dst_samples)
+{
+    pcm16_to_xlaw(linear_to_ulaw, src_length, src_samples, dst_samples);
+}
+
+function alaw_to_pcm16(src_length, src_samples, dst_samples)
+{
+    xlaw_to_pcm16(alaw_to_linear, src_length, src_samples, dst_samples);
+}
+
+function ulaw_to_pcm16(src_length, src_samples, dst_samples)
+{
+    xlaw_to_pcm16(ulaw_to_linear, src_length, src_samples, dst_samples);
+}
+
+function pcm16_alaw_tableinit()
+{
+    build_linear_to_xlaw_table(linear_to_alaw, linear2alaw);
+}
+
+function pcm16_ulaw_tableinit()
+{
+    build_linear_to_xlaw_table(linear_to_ulaw, linear2ulaw);
+}
+
+function alaw_pcm16_tableinit()
+{
+    build_xlaw_to_linear_table(alaw_to_linear, alaw2linear);
+}
+
+function ulaw_pcm16_tableinit()
+{
+    build_xlaw_to_linear_table(ulaw_to_linear, ulaw2linear);
+}
+
+for(var i =0; i < 65536;i++){
+    short_index[i] = i;
+}
+
+pcm16_alaw_tableinit();
+pcm16_ulaw_tableinit();
+alaw_pcm16_tableinit();
+ulaw_pcm16_tableinit();
--- a/web/rtc.html
+++ b/web/rtc.html