worker主要类创建和管理参数 

1. RTC::Shared 注册全局请求和通知的对象

1.创建shared
this->shared = new RTC::Shared(
/*channelMessageRegistrator*/ new ChannelMessageRegistrator(),
/*channelNotifier*/ new Channel::ChannelNotifier(this->channel));

2.channelMessageRegistrator注册request notification
this->shared->channelMessageRegistrator->RegisterHandler(
this->id,
/*channelRequestHandler*/ this,
/*channelNotificationHandler*/ this);
HandleRequest(Channel::ChannelRequest* request)


3.使用notification
auto dtlsStateChangeOffset = FBS::WebRtcTransport::CreateDtlsStateChangeNotification(

this->shared->channelNotifier->GetBufferBuilder(), FBS::WebRtcTransport::DtlsState::CLOSED);

this->shared->channelNotifier->Emit(
this->id,
FBS::Notification::Event::WEBRTCTRANSPORT_DTLS_STATE_CHANGE,
FBS::Notification::Body::WebRtcTransport_DtlsStateChangeNotification,
dtlsStateChangeOffset);

HandleNotification(Channel::ChannelNotification* notification)

2. RTC::WebRtcServer 创建webrtc服务

a.封装底层 WebRTC ICE + DTLS + UDP/TCP 套接字的监听和接入
b.提供连接能力（收包、收握手）
c.是 Transport 的“网口”

// c++
auto* webRtcServer = new RTC::WebRtcServer(this->shared, webRtcServerId, body->listenInfos());
this->mapWebRtcServers[webRtcServerId] = webRtcServer;

// 关联参数
static const char * const names[] = {
    "webRtcServerId",
    "listenInfos"
};

static const char * const names[] = {
    "protocol",
    "ip",
    "announcedAddress",
    "port",
    "portRange",
    "flags",
    "sendBufferSize",
    "recvBufferSize"
};

3. RTC::Router

a.是媒体路由核心，管理 RTP 编解码能力、Producer、Consumer
b.用于同一“房间”中的用户媒体流转发与桥接
c.一个 Router 表示一个独立的 RTP 空间


// c++
auto* router = new RTC::Router(this->shared, routerId, this);
this->mapRouters[routerId] = router;


// 关联参数
static const char * const names[] = {
    "routerId"
};

4. RTC::WebRtcTransport

a.接收远端客户端的 WebRTC ICE 候选（ufrag/password）
b.与远端协商 DTLS，然后收发 RTP/RTCP
c.挂载 Producer/Consumer 用于音视频转发

// c++ 绑定webRtcServer()
auto iceCandidates = webRtcServer->GetIceCandidates(options->enableUdp(), options->enableTcp(), options->preferUdp() options->preferTcp());

// This may throw.
auto* webRtcTransport = new RTC::WebRtcTransport(this->shared, transportId, this, webRtcServer, iceCandidates, options);

// Insert into the map.
this->mapTransports[transportId] = webRtcTransport;

// 关联参数
static const char * const names[] = {
    "transportId",
    "options"
};

static const char * const names[] = {
    "base",
    "listen_type",
    "listen",
    "enableUdp",
    "enableTcp",
    "preferUdp",
    "preferTcp",
    "iceConsentTimeout"
};


// ICE DTLS
// Create a ICE server.
this->iceServer = new RTC::IceServer(this, Utils::Crypto::GetRandomString(32), Utils::Crypto::GetRandomString(32), iceConsentTimeout);

// Create a DTLS transport.
this->dtlsTransport = new RTC::DtlsTransport(this);


// SRTP
// OnDtlsTransportConnected
this->srtpSendSession = new RTC::SrtpSession(RTC::SrtpSession::Type::OUTBOUND, srtpCryptoSuite, srtpLocalKey, srtpLocalKeyLen);

this->srtpRecvSession = new RTC::SrtpSession(RTC::SrtpSession::Type::INBOUND, srtpCryptoSuite, srtpRemoteKey, srtpRemoteKeyLen);

## Transport对比

类型	用途	使用网络	支持SRTP	典型使用者
WebRtcTransport	浏览器/WebRTC SDK	✅	✅	WebRTC 客户端
PlainTransport	RTP 推流工具/FFmpeg/GStreamer	✅	❌	RTP encoder/decoder
PipeTransport	Router <-> Router 内部通信	✅	❌	mediasoup 内部连接
DirectTransport	注入裸帧/本地媒体桥接	❌	❌	服务器本地 TTS、混音等

5. RTC::Producer

auto* producer = new RTC::Producer(this->shared, producerId, this, body);

// 关联参数
static const char * const names[] = {
    "producerId",
    "kind",
    "rtpParameters",
    "rtpMapping",
    "keyFrameRequestDelay",
    "paused"
};

//MediaKindTypeTable
static const char * const names[] = {
    "AUDIO",
    "VIDEO"
};

//RtpParametersTypeTable
static const char * const names[] = {
    "mid",
    "codecs",
    "headerExtensions",
    "encodings",
    "rtcp"
};

//RtpMappingTypeTable
static const char * const names[] = {
    "codecs",
    "encodings"
};

6 RTC::SimpleConsumer

consumer = new RTC::SimpleConsumer(this->shared, consumerId, producerId, this, body);

//关联参数
static const char * const names[] = {
    "consumerId",
    "producerId",
    "kind",
    "rtpParameters",
    "type",
    "consumableRtpEncodings",
    "paused",
    "preferredLayers",
    "ignoreDtx"
};

//MediaKindTypeTable
static const char * const names[] = {
    "AUDIO",
    "VIDEO"
};

//RtpParametersTypeTable
static const char * const names[] = {
    "mid",
    "codecs",
    "headerExtensions",
    "encodings",
    "rtcp"
};

//TypeTypeTable
static const char * const names[] = {
    "SIMPLE",
    "SIMULCAST",
    "SVC",
    "PIPE"
};

//RtpEncodingParametersTypeTable
static const char * const names[] = {
    "ssrc",
    "rid",
    "codecPayloadType",
    "rtx",
    "dtx",
    "scalabilityMode",
    "maxBitrate"
};

//ConsumerLayersTypeTable
static const char * const names[] = {
    "spatialLayer",
    "temporalLayer"
};

7. RTC::RtpStreamRecv

RTC::RtpStream::Params params;
params.encodingIdx    = encodingIdx;
params.ssrc           = ssrc;
params.payloadType    = mediaCodec.payloadType;
params.mimeType       = mediaCodec.mimeType;
params.clockRate      = mediaCodec.clockRate;
params.rid            = encoding.rid;
params.cname          = this->rtpParameters.rtcp.cname;
params.spatialLayers  = encoding.spatialLayers;
params.temporalLayers = encoding.temporalLayers;

auto* rtpStream = new RTC::RtpStreamRecv(this, params, SendNackDelay, useRtpInactivityCheck);

8. RtpStreamSend

RTC::RtpStream::Params params;
params.ssrc        = encoding.ssrc;
params.payloadType = mediaCodec->payloadType;
params.mimeType    = mediaCodec->mimeType;
params.clockRate   = mediaCodec->clockRate;
params.cname       = this->rtpParameters.rtcp.cname;

this->rtpStream = new RTC::RtpStreamSend(this, params, this->rtpParameters.mid);