58. WebRTC在Android中的应用实战

58. WebRTC在Android中的应用实战

摘要

本文深入解析WebRTC在Android智能安防系统中的应用实践，涵盖WebRTC架构原理、Android SDK集成、音视频采集渲染、信令交换、连接管理以及性能优化策略。通过某智能门铃项目的真实案例，展示如何基于WebRTC实现低延迟(<100ms)的实时音视频通信，包含完整的Android端实现代码和最佳实践总结。

关键词: WebRTC, Android, 实时音视频, PeerConnection, 智能安防, 低延迟通信

一、WebRTC架构概览

1.1 WebRTC核心组件

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应用层 API

PeerConnection API

Session Management

Media Stream API

ICE/STUN/TURN

Audio Engine

Video Engine

NetEQ 抗丢包

AEC 回声消除

AGC 增益控制

VP8/VP9/H264

Jitter Buffer

Image Enhancement

Network Layer

核心模块说明：

1.2 WebRTC通信流程

设备B信令服务器设备A设备B信令服务器设备A#mermaid-svg-lQRD1vNaKT7VUlSc{font-family:\”trebuchet ms\”,verdana,arial,sans-serif;font-size:16px;fill:#333;}@keyframes edge-animation-frame{from{stroke-dashoffset:0;}}@keyframes dash{to{stroke-dashoffset:0;}}#mermaid-svg-lQRD1vNaKT7VUlSc .edge-animation-slow{stroke-dasharray:9,5!important;stroke-dashoffset:900;animation:dash 50s linear infinite;stroke-linecap:round;}#mermaid-svg-lQRD1vNaKT7VUlSc .edge-animation-fast{stroke-dasharray:9,5!important;stroke-dashoffset:900;animation:dash 20s linear infinite;stroke-linecap:round;}#mermaid-svg-lQRD1vNaKT7VUlSc .error-icon{fill:#552222;}#mermaid-svg-lQRD1vNaKT7VUlSc .error-text{fill:#552222;stroke:#552222;}#mermaid-svg-lQRD1vNaKT7VUlSc .edge-thickness-normal{stroke-width:1px;}#mermaid-svg-lQRD1vNaKT7VUlSc .edge-thickness-thick{stroke-width:3.5px;}#mermaid-svg-lQRD1vNaKT7VUlSc .edge-pattern-solid{stroke-dasharray:0;}#mermaid-svg-lQRD1vNaKT7VUlSc .edge-thickness-invisible{stroke-width:0;fill:none;}#mermaid-svg-lQRD1vNaKT7VUlSc .edge-pattern-dashed{stroke-dasharray:3;}#mermaid-svg-lQRD1vNaKT7VUlSc .edge-pattern-dotted{stroke-dasharray:2;}#mermaid-svg-lQRD1vNaKT7VUlSc .marker{fill:#333333;stroke:#333333;}#mermaid-svg-lQRD1vNaKT7VUlSc .marker.cross{stroke:#333333;}#mermaid-svg-lQRD1vNaKT7VUlSc svg{font-family:\”trebuchet ms\”,verdana,arial,sans-serif;font-size:16px;}#mermaid-svg-lQRD1vNaKT7VUlSc p{margin:0;}#mermaid-svg-lQRD1vNaKT7VUlSc .actor{stroke:hsl(259.6261682243, 59.7765363128%, 87.9019607843%);fill:#ECECFF;}#mermaid-svg-lQRD1vNaKT7VUlSc text.actor>tspan{fill:black;stroke:none;}#mermaid-svg-lQRD1vNaKT7VUlSc .actor-line{stroke:hsl(259.6261682243, 59.7765363128%, 87.9019607843%);}#mermaid-svg-lQRD1vNaKT7VUlSc .innerArc{stroke-width:1.5;stroke-dasharray:none;}#mermaid-svg-lQRD1vNaKT7VUlSc .messageLine0{stroke-width:1.5;stroke-dasharray:none;stroke:#333;}#mermaid-svg-lQRD1vNaKT7VUlSc .messageLine1{stroke-width:1.5;stroke-dasharray:2,2;stroke:#333;}#mermaid-svg-lQRD1vNaKT7VUlSc #arrowhead path{fill:#333;stroke:#333;}#mermaid-svg-lQRD1vNaKT7VUlSc .sequenceNumber{fill:white;}#mermaid-svg-lQRD1vNaKT7VUlSc #sequencenumber{fill:#333;}#mermaid-svg-lQRD1vNaKT7VUlSc #crosshead path{fill:#333;stroke:#333;}#mermaid-svg-lQRD1vNaKT7VUlSc .messageText{fill:#333;stroke:none;}#mermaid-svg-lQRD1vNaKT7VUlSc .labelBox{stroke:hsl(259.6261682243, 59.7765363128%, 87.9019607843%);fill:#ECECFF;}#mermaid-svg-lQRD1vNaKT7VUlSc .labelText,#mermaid-svg-lQRD1vNaKT7VUlSc .labelText>tspan{fill:black;stroke:none;}#mermaid-svg-lQRD1vNaKT7VUlSc .loopText,#mermaid-svg-lQRD1vNaKT7VUlSc .loopText>tspan{fill:black;stroke:none;}#mermaid-svg-lQRD1vNaKT7VUlSc .loopLine{stroke-width:2px;stroke-dasharray:2,2;stroke:hsl(259.6261682243, 59.7765363128%, 87.9019607843%);fill:hsl(259.6261682243, 59.7765363128%, 87.9019607843%);}#mermaid-svg-lQRD1vNaKT7VUlSc .note{stroke:#aaaa33;fill:#fff5ad;}#mermaid-svg-lQRD1vNaKT7VUlSc .noteText,#mermaid-svg-lQRD1vNaKT7VUlSc .noteText>tspan{fill:black;stroke:none;}#mermaid-svg-lQRD1vNaKT7VUlSc .activation0{fill:#f4f4f4;stroke:#666;}#mermaid-svg-lQRD1vNaKT7VUlSc .activation1{fill:#f4f4f4;stroke:#666;}#mermaid-svg-lQRD1vNaKT7VUlSc .activation2{fill:#f4f4f4;stroke:#666;}#mermaid-svg-lQRD1vNaKT7VUlSc .actorPopupMenu{position:absolute;}#mermaid-svg-lQRD1vNaKT7VUlSc .actorPopupMenuPanel{position:absolute;fill:#ECECFF;box-shadow:0px 8px 16px 0px rgba(0,0,0,0.2);filter:drop-shadow(3px 5px 2px rgb(0 0 0 / 0.4));}#mermaid-svg-lQRD1vNaKT7VUlSc .actor-man line{stroke:hsl(259.6261682243, 59.7765363128%, 87.9019607843%);fill:#ECECFF;}#mermaid-svg-lQRD1vNaKT7VUlSc .actor-man circle,#mermaid-svg-lQRD1vNaKT7VUlSc line{stroke:hsl(259.6261682243, 59.7765363128%, 87.9019607843%);fill:#ECECFF;stroke-width:2px;}#mermaid-svg-lQRD1vNaKT7VUlSc :root{–mermaid-font-family:\”trebuchet ms\”,verdana,arial,sans-serif;}ICE候选交换建立P2P连接创建PeerConnection添加本地媒体流创建Offer发送Offer转发Offer创建PeerConnection设置RemoteDescription创建Answer发送Answer转发Answer设置RemoteDescription发送ICE Candidate转发ICE Candidate发送ICE Candidate转发ICE Candidate实时音视频数据实时音视频数据

二、Android WebRTC SDK集成

2.1 依赖配置

// build.gradle (Project)
buildscript {
repositories {
google()
mavenCentral()
}
}

// build.gradle (Module: app)
dependencies {
// WebRTC官方库
implementation 'org.webrtc:google-webrtc:1.0.32006'

// 信令通信
implementation 'com.squareup.okhttp3:okhttp:4.10.0'
implementation 'com.google.code.gson:gson:2.10'

// 协程
implementation 'org.jetbrains.kotlinx:kotlinx-coroutines-android:1.6.4'

// 权限管理
implementation 'com.guolindev.permissionx:permissionx:1.7.1'
}

// AndroidManifest.xml
<uses-permission android:name="android.permission.INTERNET" />
<uses-permission android:name="android.permission.CAMERA" />
<uses-permission android:name="android.permission.RECORD_AUDIO" />
<uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
<uses-permission android:name="android.permission.ACCESS_NETWORK_STATE" />
<uses-permission android:name="android.permission.CHANGE_NETWORK_STATE" />

2.2 WebRTC初始化

/**
* WebRTC管理器
* 负责WebRTC的初始化和生命周期管理
*/
class WebRtcManager(private val context: Context) {
private var peerConnectionFactory: PeerConnectionFactory? = null
private var audioSource: AudioSource? = null
private var videoSource: VideoSource? = null
private var videoCapturer: VideoCapturer? = null

companion object {
private const val TAG = "WebRtcManager"

// 音频约束
private const val AUDIO_ECHO_CANCELLATION = "googEchoCancellation"
private const val AUDIO_AUTO_GAIN_CONTROL = "googAutoGainControl"
private const val AUDIO_HIGH_PASS_FILTER = "googHighpassFilter"
private const val AUDIO_NOISE_SUPPRESSION = "googNoiseSuppression"
}

/**
* 初始化WebRTC
*/
fun initialize() {
// 初始化PeerConnectionFactory参数
val initializationOptions = PeerConnectionFactory.InitializationOptions.builder(context)
.setEnableInternalTracer(false)
.setFieldTrials("") // 可选的实验性特性
.createInitializationOptions()

PeerConnectionFactory.initialize(initializationOptions)

// 创建PeerConnectionFactory
val options = PeerConnectionFactory.Options().apply {
// 禁用网络忽略掩码（允许使用所有网络接口）
networkIgnoreMask = 0
}

val encoderFactory = DefaultVideoEncoderFactory(
EglBase.create().eglBaseContext,
true, // enableIntelVp8Encoder
true // enableH264HighProfile
)

val decoderFactory = DefaultVideoDecoderFactory(
EglBase.create().eglBaseContext
)

peerConnectionFactory = PeerConnectionFactory.builder()
.setOptions(options)
.setVideoEncoderFactory(encoderFactory)
.setVideoDecoderFactory(decoderFactory)
.createPeerConnectionFactory()

Log.i(TAG, "WebRTC initialized successfully")
}

/**
* 创建音频源
*/
fun createAudioSource(): AudioSource {
if (audioSource == null) {
val audioConstraints = MediaConstraints().apply {
mandatory.add(MediaConstraints.KeyValuePair(AUDIO_ECHO_CANCELLATION, "true"))
mandatory.add(MediaConstraints.KeyValuePair(AUDIO_AUTO_GAIN_CONTROL, "true"))
mandatory.add(MediaConstraints.KeyValuePair(AUDIO_HIGH_PASS_FILTER, "true"))
mandatory.add(MediaConstraints.KeyValuePair(AUDIO_NOISE_SUPPRESSION, "true"))
}

audioSource = peerConnectionFactory?.createAudioSource(audioConstraints)
Log.i(TAG, "Audio source created")
}

return audioSource!!
}

/**
* 创建视频源
*/
fun createVideoSource(isScreencast: Boolean = false): VideoSource {
if (videoSource == null) {
videoSource = peerConnectionFactory?.createVideoSource(isScreencast)
Log.i(TAG, "Video source created")
}

return videoSource!!
}

/**
* 创建音频轨道
*/
fun createAudioTrack(trackId: String = "audio_track"): AudioTrack {
val audioSource = createAudioSource()
return peerConnectionFactory!!.createAudioTrack(trackId, audioSource)
}

/**
* 创建视频轨道
*/
fun createVideoTrack(trackId: String = "video_track"): VideoTrack {
val videoSource = createVideoSource()
return peerConnectionFactory!!.createVideoTrack(trackId, videoSource)
}

/**
* 创建PeerConnection
*/
fun createPeerConnection(
iceServers: List<PeerConnection.IceServer>,
observer: PeerConnection.Observer
): PeerConnection? {
val rtcConfig = PeerConnection.RTCConfiguration(iceServers).apply {
// TCP候选策略
tcpCandidatePolicy = PeerConnection.TcpCandidatePolicy.ENABLED

// Bundle策略
bundlePolicy = PeerConnection.BundlePolicy.MAXBUNDLE

// RTCP复用策略
rtcpMuxPolicy = PeerConnection.RtcpMuxPolicy.REQUIRE

// 连续收集ICE候选
continualGatheringPolicy = PeerConnection.ContinualGatheringPolicy.GATHER_CONTINUALLY

// 启用DTLS
enableDtlsSrtp = true

// SDP语义
sdpSemantics = PeerConnection.SdpSemantics.UNIFIED_PLAN
}

return peerConnectionFactory?.createPeerConnection(rtcConfig, observer)
}

/**
* 获取PeerConnectionFactory
*/
fun getPeerConnectionFactory(): PeerConnectionFactory? = peerConnectionFactory

/**
* 释放资源
*/
fun dispose() {
videoCapturer?.dispose()
videoSource?.dispose()
audioSource?.dispose()
peerConnectionFactory?.dispose()

videoCapturer = null
videoSource = null
audioSource = null
peerConnectionFactory = null

Log.i(TAG, "WebRTC resources disposed")
}
}

三、音视频采集与渲染

3.1 摄像头采集

/**
* 摄像头采集器
*/
class CameraCapturerManager(
private val context: Context,
private val webRtcManager: WebRtcManager
) {
private var videoCapturer: CameraVideoCapturer? = null
private var surfaceTextureHelper: SurfaceTextureHelper? = null

companion object {
private const val TAG = "CameraCapturer"
private const val VIDEO_WIDTH = 1280
private const val VIDEO_HEIGHT = 720
private const val VIDEO_FPS = 30
}

/**
* 初始化摄像头
*/
fun initialize(): Boolean {
// 创建Camera1或Camera2采集器
videoCapturer = createCameraVideoCapturer()

if (videoCapturer == null) {
Log.e(TAG, "Failed to create camera capturer")
return false
}

// 创建SurfaceTextureHelper
val eglBase = EglBase.create()
surfaceTextureHelper = SurfaceTextureHelper.create(
"CaptureThread",
eglBase.eglBaseContext
)

// 初始化采集器
val videoSource = webRtcManager.createVideoSource()
videoCapturer?.initialize(
surfaceTextureHelper,
context,
videoSource.capturerObserver
)

Log.i(TAG, "Camera capturer initialized")
return true
}

/**
* 创建摄像头采集器（优先使用Camera2）
*/
private fun createCameraVideoCapturer(): CameraVideoCapturer? {
val enumerator = if (Camera2Enumerator.isSupported(context)) {
Camera2Enumerator(context)
} else {
Camera1Enumerator(true)
}

// 优先使用前置摄像头
val deviceNames = enumerator.deviceNames
for (deviceName in deviceNames) {
if (enumerator.isFrontFacing(deviceName)) {
val capturer = enumerator.createCapturer(deviceName, null)
if (capturer != null) {
Log.i(TAG, "Using front camera: $deviceName")
return capturer
}
}
}

// 使用后置摄像头
for (deviceName in deviceNames) {
if (enumerator.isBackFacing(deviceName)) {
val capturer = enumerator.createCapturer(deviceName, null)
if (capturer != null) {
Log.i(TAG, "Using back camera: $deviceName")
return capturer
}
}
}

return null
}

/**
* 开始采集
*/
fun startCapture() {
videoCapturer?.startCapture(VIDEO_WIDTH, VIDEO_HEIGHT, VIDEO_FPS)
Log.i(TAG, "Camera capture started: ${VIDEO_WIDTH}x${VIDEO_HEIGHT}@${VIDEO_FPS}fps")
}

/**
* 停止采集
*/
fun stopCapture() {
try {
videoCapturer?.stopCapture()
Log.i(TAG, "Camera capture stopped")
} catch (e: InterruptedException) {
Log.e(TAG, "Failed to stop capture", e)
}
}

/**
* 切换摄像头
*/
fun switchCamera(switchHandler: CameraVideoCapturer.CameraSwitchHandler) {
if (videoCapturer is CameraVideoCapturer) {
(videoCapturer as CameraVideoCapturer).switchCamera(switchHandler)
}
}

/**
* 释放资源
*/
fun dispose() {
videoCapturer?.dispose()
surfaceTextureHelper?.dispose()

videoCapturer = null
surfaceTextureHelper = null

Log.i(TAG, "Camera capturer disposed")
}
}

3.2 视频渲染

/**
* 视频渲染管理器
*/
class VideoRendererManager(
private val context: Context
) {
private var localRenderer: SurfaceViewRenderer? = null
private var remoteRenderer: SurfaceViewRenderer? = null
private val eglBase = EglBase.create()

companion object {
private const val TAG = "VideoRenderer"
}

/**
* 初始化本地渲染器
*/
fun initializeLocalRenderer(surfaceView: SurfaceViewRenderer): SurfaceViewRenderer {
surfaceView.init(eglBase.eglBaseContext, null)
surfaceView.setMirror(true) // 镜像显示本地视频
surfaceView.setEnableHardwareScaler(true)
surfaceView.setZOrderMediaOverlay(true)

localRenderer = surfaceView
Log.i(TAG, "Local renderer initialized")

return surfaceView
}

/**
* 初始化远端渲染器
*/
fun initializeRemoteRenderer(surfaceView: SurfaceViewRenderer): SurfaceViewRenderer {
surfaceView.init(eglBase.eglBaseContext, null)
surfaceView.setMirror(false)
surfaceView.setEnableHardwareScaler(true)

remoteRenderer = surfaceView
Log.i(TAG, "Remote renderer initialized")

return surfaceView
}

/**
* 添加本地视频轨道
*/
fun addLocalVideoTrack(videoTrack: VideoTrack) {
localRenderer?.let { renderer ->
videoTrack.addSink(renderer)
Log.i(TAG, "Local video track added")
}
}

/**
* 添加远端视频轨道
*/
fun addRemoteVideoTrack(videoTrack: VideoTrack) {
remoteRenderer?.let { renderer ->
videoTrack.addSink(renderer)
Log.i(TAG, "Remote video track added")
}
}

/**
* 移除本地视频轨道
*/
fun removeLocalVideoTrack(videoTrack: VideoTrack) {
localRenderer?.let { renderer ->
videoTrack.removeSink(renderer)
Log.i(TAG, "Local video track removed")
}
}

/**
* 移除远端视频轨道
*/
fun removeRemoteVideoTrack(videoTrack: VideoTrack) {
remoteRenderer?.let { renderer ->
videoTrack.removeSink(renderer)
Log.i(TAG, "Remote video track removed")
}
}

/**
* 释放资源
*/
fun dispose() {
localRenderer?.release()
remoteRenderer?.release()
eglBase.release()

localRenderer = null
remoteRenderer = null

Log.i(TAG, "Video renderers disposed")
}
}

四、PeerConnection管理

4.1 完整的PeerConnection封装

/**
* WebRTC PeerConnection封装
*/
class WebRtcPeerConnection(
private val webRtcManager: WebRtcManager,
private val iceServers: List<PeerConnection.IceServer>,
private val listener: PeerConnectionListener
) {
private var peerConnection: PeerConnection? = null
private var localMediaStream: MediaStream? = null
private val queuedRemoteCandidates = mutableListOf<IceCandidate>()
private var isRemoteDescriptionSet = false

companion object {
private const val TAG = "WebRtcPeerConnection"
private const val STREAM_ID = "stream_id"
private const val AUDIO_TRACK_ID = "audio_track"
private const val VIDEO_TRACK_ID = "video_track"
}

private val peerConnectionObserver = object : PeerConnection.Observer {
override fun onSignalingChange(newState: PeerConnection.SignalingState?) {
Log.d(TAG, "onSignalingChange: $newState")
}

override fun onIceConnectionChange(newState: PeerConnection.IceConnectionState?) {
Log.d(TAG, "onIceConnectionChange: $newState")
newState?.let {
when (it) {
PeerConnection.IceConnectionState.CONNECTED -> {
listener.onConnectionStateChanged(ConnectionState.CONNECTED)
}
PeerConnection.IceConnectionState.DISCONNECTED -> {
listener.onConnectionStateChanged(ConnectionState.DISCONNECTED)
}
PeerConnection.IceConnectionState.FAILED -> {
listener.onConnectionStateChanged(ConnectionState.FAILED)
}
else -> {}
}
}
}

override fun onIceConnectionReceivingChange(receiving: Boolean) {
Log.d(TAG, "onIceConnectionReceivingChange: $receiving")
}

override fun onIceGatheringChange(newState: PeerConnection.IceGatheringState?) {
Log.d(TAG, "onIceGatheringChange: $newState")
}

override fun onIceCandidate(candidate: IceCandidate?) {
Log.d(TAG, "onIceCandidate: ${candidate?.sdp}")
candidate?.let {
listener.onIceCandidate(it)
}
}

override fun onIceCandidatesRemoved(candidates: Array<out IceCandidate>?) {
Log.d(TAG, "onIceCandidatesRemoved: ${candidates?.size}")
}

override fun onAddStream(stream: MediaStream?) {
Log.d(TAG, "onAddStream: ${stream?.id}")
stream?.let {
if (it.videoTracks.isNotEmpty()) {
listener.onRemoteVideoTrack(it.videoTracks[0])
}
if (it.audioTracks.isNotEmpty()) {
listener.onRemoteAudioTrack(it.audioTracks[0])
}
}
}

override fun onRemoveStream(stream: MediaStream?) {
Log.d(TAG, "onRemoveStream: ${stream?.id}")
}

override fun onDataChannel(dataChannel: DataChannel?) {
Log.d(TAG, "onDataChannel: ${dataChannel?.label()}")
}

override fun onRenegotiationNeeded() {
Log.d(TAG, "onRenegotiationNeeded")
}

override fun onAddTrack(receiver: RtpReceiver?, streams: Array<out MediaStream>?) {
Log.d(TAG, "onAddTrack: ${receiver?.track()?.kind()}")
}
}

/**
* 初始化PeerConnection
*/
fun initialize(): Boolean {
peerConnection = webRtcManager.createPeerConnection(iceServers, peerConnectionObserver)

if (peerConnection == null) {
Log.e(TAG, "Failed to create PeerConnection")
return false
}

Log.i(TAG, "PeerConnection initialized")
return true
}

/**
* 添加本地媒体流
*/
fun addLocalMediaStream(hasAudio: Boolean = true, hasVideo: Boolean = true) {
val factory = webRtcManager.getPeerConnectionFactory() ?: return

localMediaStream = factory.createLocalMediaStream(STREAM_ID)

if (hasAudio) {
val audioTrack = webRtcManager.createAudioTrack(AUDIO_TRACK_ID)
localMediaStream?.addTrack(audioTrack)
Log.i(TAG, "Local audio track added")
}

if (hasVideo) {
val videoTrack = webRtcManager.createVideoTrack(VIDEO_TRACK_ID)
localMediaStream?.addTrack(videoTrack)
listener.onLocalVideoTrack(videoTrack)
Log.i(TAG, "Local video track added")
}

peerConnection?.addStream(localMediaStream)
Log.i(TAG, "Local media stream added to PeerConnection")
}

/**
* 创建Offer
*/
fun createOffer() {
val constraints = MediaConstraints().apply {
mandatory.add(MediaConstraints.KeyValuePair("OfferToReceiveAudio", "true"))
mandatory.add(MediaConstraints.KeyValuePair("OfferToReceiveVideo", "true"))
}

peerConnection?.createOffer(object : SdpObserver {
override fun onCreateSuccess(sessionDescription: SessionDescription?) {
Log.i(TAG, "Create offer success")
sessionDescription?.let {
peerConnection?.setLocalDescription(object : SdpObserver {
override fun onSetSuccess() {
Log.i(TAG, "Set local description success")
listener.onLocalSessionDescription(it)
}

override fun onSetFailure(error: String?) {
Log.e(TAG, "Set local description failed: $error")
}

override fun onCreateSuccess(p0: SessionDescription?) {}
override fun onCreateFailure(p0: String?) {}
}, it)
}
}

override fun onSetSuccess() {}

override fun onCreateFailure(error: String?) {
Log.e(TAG, "Create offer failed: $error")
}

override fun onSetFailure(error: String?) {}
}, constraints)
}

/**
* 创建Answer
*/
fun createAnswer() {
val constraints = MediaConstraints().apply {
mandatory.add(MediaConstraints.KeyValuePair("OfferToReceiveAudio", "true"))
mandatory.add(MediaConstraints.KeyValuePair("OfferToReceiveVideo", "true"))
}

peerConnection?.createAnswer(object : SdpObserver {
override fun onCreateSuccess(sessionDescription: SessionDescription?) {
Log.i(TAG, "Create answer success")
sessionDescription?.let {
peerConnection?.setLocalDescription(object : SdpObserver {
override fun onSetSuccess() {
Log.i(TAG, "Set local description success")
listener.onLocalSessionDescription(it)
}

override fun onSetFailure(error: String?) {
Log.e(TAG, "Set local description failed: $error")
}

override fun onCreateSuccess(p0: SessionDescription?) {}
override fun onCreateFailure(p0: String?) {}
}, it)
}
}

override fun onSetSuccess() {}

override fun onCreateFailure(error: String?) {
Log.e(TAG, "Create answer failed: $error")
}

override fun onSetFailure(error: String?) {}
}, constraints)
}

/**
* 设置远端SDP
*/
fun setRemoteDescription(sessionDescription: SessionDescription) {
peerConnection?.setRemoteDescription(object : SdpObserver {
override fun onSetSuccess() {
Log.i(TAG, "Set remote description success")
isRemoteDescriptionSet = true

// 添加缓存的ICE候选
queuedRemoteCandidates.forEach { candidate ->
addIceCandidate(candidate)
}
queuedRemoteCandidates.clear()
}

override fun onSetFailure(error: String?) {
Log.e(TAG, "Set remote description failed: $error")
}

override fun onCreateSuccess(p0: SessionDescription?) {}
override fun onCreateFailure(p0: String?) {}
}, sessionDescription)
}

/**
* 添加ICE候选
*/
fun addIceCandidate(candidate: IceCandidate) {
if (isRemoteDescriptionSet) {
peerConnection?.addIceCandidate(candidate)
Log.d(TAG, "ICE candidate added: ${candidate.sdp}")
} else {
// 缓存候选，等待远端描述设置后再添加
queuedRemoteCandidates.add(candidate)
Log.d(TAG, "ICE candidate queued (remote description not set yet)")
}
}

/**
* 关闭连接
*/
fun close() {
peerConnection?.close()
peerConnection = null
localMediaStream = null
queuedRemoteCandidates.clear()
isRemoteDescriptionSet = false

Log.i(TAG, "PeerConnection closed")
}

/**
* 获取连接统计信息
*/
fun getStats(callback: RTCStatsCollectorCallback) {
peerConnection?.getStats(callback)
}
}

/**
* PeerConnection监听器
*/
interface PeerConnectionListener {
fun onLocalSessionDescription(sdp: SessionDescription)
fun onIceCandidate(candidate: IceCandidate)
fun onLocalVideoTrack(videoTrack: VideoTrack)
fun onRemoteVideoTrack(videoTrack: VideoTrack)
fun onRemoteAudioTrack(audioTrack: AudioTrack)
fun onConnectionStateChanged(state: ConnectionState)
}

enum class ConnectionState {
NEW,
CONNECTING,
CONNECTED,
DISCONNECTED,
FAILED,
CLOSED
}

五、信令服务器通信

5.1 WebSocket信令客户端

/**
* WebSocket信令客户端
*/
class WebSocketSignalingClient(
private val serverUrl: String,
private val roomId: String,
private val userId: String
) {
private var webSocket: WebSocket? = null
private val client = OkHttpClient.Builder()
.readTimeout(30, TimeUnit.SECONDS)
.writeTimeout(30, TimeUnit.SECONDS)
.pingInterval(20, TimeUnit.SECONDS)
.build()

private var listener: SignalingListener? = null

companion object {
private const val TAG = "SignalingClient"
}

/**
* 连接信令服务器
*/
fun connect(listener: SignalingListener) {
this.listener = listener

val request = Request.Builder()
.url("$serverUrl?roomId=$roomId&userId=$userId")
.build()

webSocket = client.newWebSocket(request, object : WebSocketListener() {
override fun onOpen(webSocket: WebSocket, response: Response) {
Log.i(TAG, "WebSocket connected")
listener.onConnected()

// 发送加入房间消息
sendJoinRoom()
}

override fun onMessage(webSocket: WebSocket, text: String) {
Log.d(TAG, "Message received: $text")
handleMessage(text)
}

override fun onClosing(webSocket: WebSocket, code: Int, reason: String) {
Log.i(TAG, "WebSocket closing: $code – $reason")
}

override fun onClosed(webSocket: WebSocket, code: Int, reason: String) {
Log.i(TAG, "WebSocket closed: $code – $reason")
listener.onDisconnected()
}

override fun onFailure(webSocket: WebSocket, t: Throwable, response: Response?) {
Log.e(TAG, "WebSocket error", t)
listener.onError(t.message ?: "Unknown error")
}
})
}

/**
* 发送加入房间消息
*/
private fun sendJoinRoom() {
val message = mapOf(
"type" to "join",
"roomId" to roomId,
"userId" to userId
)
sendMessage(message)
}

/**
* 发送Offer
*/
fun sendOffer(sdp: String) {
val message = mapOf(
"type" to "offer",
"roomId" to roomId,
"userId" to userId,
"sdp" to sdp
)
sendMessage(message)
}

/**
* 发送Answer
*/
fun sendAnswer(sdp: String) {
val message = mapOf(
"type" to "answer",
"roomId" to roomId,
"userId" to userId,
"sdp" to sdp
)
sendMessage(message)
}

/**
* 发送ICE候选
*/
fun sendIceCandidate(candidate: IceCandidate) {
val message = mapOf(
"type" to "candidate",
"roomId" to roomId,
"userId" to userId,
"candidate" to mapOf(
"sdpMid" to candidate.sdpMid,
"sdpMLineIndex" to candidate.sdpMLineIndex,
"sdp" to candidate.sdp
)
)
sendMessage(message)
}

/**
* 发送消息
*/
private fun sendMessage(message: Map<String, Any>) {
val json = Gson().toJson(message)
webSocket?.send(json)
Log.d(TAG, "Message sent: $json")
}

/**
* 处理接收到的消息
*/
private fun handleMessage(text: String) {
try {
val json = Gson().fromJson(text, Map::class.java)
val type = json["type"] as? String

when (type) {
"joined" -> {
val users = json["users"] as? List<String>
listener?.onRoomJoined(users ?: emptyList())
}
"user-joined" -> {
val newUserId = json["userId"] as? String
newUserId?.let { listener?.onUserJoined(it) }
}
"offer" -> {
val sdp = json["sdp"] as? String
sdp?.let { listener?.onOfferReceived(it) }
}
"answer" -> {
val sdp = json["sdp"] as? String
sdp?.let { listener?.onAnswerReceived(it) }
}
"candidate" -> {
val candidateData = json["candidate"] as? Map<*, *>
if (candidateData != null) {
val candidate = IceCandidate(
candidateData["sdpMid"] as String,
(candidateData["sdpMLineIndex"] as Double).toInt(),
candidateData["sdp"] as String
)
listener?.onIceCandidateReceived(candidate)
}
}
"error" -> {
val error = json["message"] as? String
listener?.onError(error ?: "Unknown error")
}
}
} catch (e: Exception) {
Log.e(TAG, "Failed to parse message", e)
}
}

/**
* 断开连接
*/
fun disconnect() {
webSocket?.close(1000, "Client closed")
webSocket = null
listener = null
}
}

/**
* 信令监听器
*/
interface SignalingListener {
fun onConnected()
fun onDisconnected()
fun onRoomJoined(users: List<String>)
fun onUserJoined(userId: String)
fun onOfferReceived(sdp: String)
fun onAnswerReceived(sdp: String)
fun onIceCandidateReceived(candidate: IceCandidate)
fun onError(error: String)
}

六、完整的通话管理器

6.1 WebRTC通话管理

/**
* WebRTC通话管理器
* 整合所有组件，提供统一的通话接口
*/
class WebRtcCallManager(
private val context: Context,
private val signalingServerUrl: String
) {
private val webRtcManager = WebRtcManager(context)
private val cameraCapturerManager = CameraCapturerManager(context, webRtcManager)
private val videoRendererManager = VideoRendererManager(context)

private var peerConnection: WebRtcPeerConnection? = null
private var signalingClient: WebSocketSignalingClient? = null

private var localVideoTrack: VideoTrack? = null
private var remoteVideoTrack: VideoTrack? = null

private var callListener: CallListener? = null

companion object {
private const val TAG = "WebRtcCallManager"

// STUN/TURN服务器配置
private val ICE_SERVERS = listOf(
PeerConnection.IceServer.builder("stun:stun.l.google.com:19302").create(),
PeerConnection.IceServer.builder("stun:stun1.l.google.com:19302").create()
)
}

/**
* 初始化
*/
fun initialize() {
webRtcManager.initialize()
cameraCapturerManager.initialize()
Log.i(TAG, "Call manager initialized")
}

/**
* 开始通话（主叫方）
*/
fun startCall(
roomId: String,
userId: String,
localVideoView: SurfaceViewRenderer,
remoteVideoView: SurfaceViewRenderer,
listener: CallListener
) {
this.callListener = listener

// 初始化渲染器
videoRendererManager.initializeLocalRenderer(localVideoView)
videoRendererManager.initializeRemoteRenderer(remoteVideoView)

// 创建PeerConnection
createPeerConnection()

// 添加本地媒体流
peerConnection?.addLocalMediaStream(hasAudio = true, hasVideo = true)

// 开始摄像头采集
cameraCapturerManager.startCapture()

// 连接信令服务器
connectSignaling(roomId, userId, isInitiator = true)
}

/**
* 接受通话（被叫方）
*/
fun acceptCall(
roomId: String,
userId: String,
localVideoView: SurfaceViewRenderer,
remoteVideoView: SurfaceViewRenderer,
listener: CallListener
) {
this.callListener = listener

// 初始化渲染器
videoRendererManager.initializeLocalRenderer(localVideoView)
videoRendererManager.initializeRemoteRenderer(remoteVideoView)

// 创建PeerConnection
createPeerConnection()

// 添加本地媒体流
peerConnection?.addLocalMediaStream(hasAudio = true, hasVideo = true)

// 开始摄像头采集
cameraCapturerManager.startCapture()

// 连接信令服务器
connectSignaling(roomId, userId, isInitiator = false)
}

/**
* 创建PeerConnection
*/
private fun createPeerConnection() {
peerConnection = WebRtcPeerConnection(
webRtcManager,
ICE_SERVERS,
peerConnectionListener
)

peerConnection?.initialize()
}

/**
* 连接信令服务器
*/
private fun connectSignaling(roomId: String, userId: String, isInitiator: Boolean) {
signalingClient = WebSocketSignalingClient(signalingServerUrl, roomId, userId)

signalingClient?.connect(object : SignalingListener {
override fun onConnected() {
Log.i(TAG, "Signaling connected")
}

override fun onDisconnected() {
Log.i(TAG, "Signaling disconnected")
callListener?.onCallEnded()
}

override fun onRoomJoined(users: List<String>) {
Log.i(TAG, "Room joined, users: $users")

if (isInitiator && users.size > 1) {
// 作为主叫方，创建Offer
peerConnection?.createOffer()
}
}

override fun onUserJoined(userId: String) {
Log.i(TAG, "User joined: $userId")

if (isInitiator) {
// 有新用户加入，创建Offer
peerConnection?.createOffer()
}
}

override fun onOfferReceived(sdp: String) {
Log.i(TAG, "Offer received")

val sessionDescription = SessionDescription(
SessionDescription.Type.OFFER,
sdp
)

peerConnection?.setRemoteDescription(sessionDescription)
peerConnection?.createAnswer()
}

override fun onAnswerReceived(sdp: String) {
Log.i(TAG, "Answer received")

val sessionDescription = SessionDescription(
SessionDescription.Type.ANSWER,
sdp
)

peerConnection?.setRemoteDescription(sessionDescription)
}

override fun onIceCandidateReceived(candidate: IceCandidate) {
Log.d(TAG, "ICE candidate received")
peerConnection?.addIceCandidate(candidate)
}

override fun onError(error: String) {
Log.e(TAG, "Signaling error: $error")
callListener?.onCallError(error)
}
})
}

/**
* PeerConnection监听器
*/
private val peerConnectionListener = object : PeerConnectionListener {
override fun onLocalSessionDescription(sdp: SessionDescription) {
Log.i(TAG, "Local session description created: ${sdp.type}")

when (sdp.type) {
SessionDescription.Type.OFFER -> {
signalingClient?.sendOffer(sdp.description)
}
SessionDescription.Type.ANSWER -> {
signalingClient?.sendAnswer(sdp.description)
}
else -> {}
}
}

override fun onIceCandidate(candidate: IceCandidate) {
Log.d(TAG, "ICE candidate generated")
signalingClient?.sendIceCandidate(candidate)
}

override fun onLocalVideoTrack(videoTrack: VideoTrack) {
Log.i(TAG, "Local video track ready")
localVideoTrack = videoTrack
videoRendererManager.addLocalVideoTrack(videoTrack)
}

override fun onRemoteVideoTrack(videoTrack: VideoTrack) {
Log.i(TAG, "Remote video track received")
remoteVideoTrack = videoTrack
videoRendererManager.addRemoteVideoTrack(videoTrack)
callListener?.onRemoteStreamReceived()
}

override fun onRemoteAudioTrack(audioTrack: AudioTrack) {
Log.i(TAG, "Remote audio track received")
}

override fun onConnectionStateChanged(state: ConnectionState) {
Log.i(TAG, "Connection state changed: $state")

when (state) {
ConnectionState.CONNECTED -> {
callListener?.onCallConnected()
}
ConnectionState.DISCONNECTED -> {
callListener?.onCallDisconnected()
}
ConnectionState.FAILED -> {
callListener?.onCallError("Connection failed")
}
else -> {}
}
}
}

/**
* 切换摄像头
*/
fun switchCamera() {
cameraCapturerManager.switchCamera(object : CameraVideoCapturer.CameraSwitchHandler {
override fun onCameraSwitchDone(isFrontCamera: Boolean) {
Log.i(TAG, "Camera switched: front=$isFrontCamera")
}

override fun onCameraSwitchError(errorDescription: String?) {
Log.e(TAG, "Camera switch error: $errorDescription")
}
})
}

/**
* 切换音频状态
*/
fun toggleAudio(enabled: Boolean) {
localVideoTrack?.setEnabled(enabled)
}

/**
* 切换视频状态
*/
fun toggleVideo(enabled: Boolean) {
localVideoTrack?.setEnabled(enabled)
}

/**
* 结束通话
*/
fun endCall() {
cameraCapturerManager.stopCapture()
peerConnection?.close()
signalingClient?.disconnect()

localVideoTrack = null
remoteVideoTrack = null

Log.i(TAG, "Call ended")
}

/**
* 释放资源
*/
fun dispose() {
endCall()
cameraCapturerManager.dispose()
videoRendererManager.dispose()
webRtcManager.dispose()

Log.i(TAG, "Call manager disposed")
}
}

/**
* 通话监听器
*/
interface CallListener {
fun onCallConnected()
fun onCallDisconnected()
fun onRemoteStreamReceived()
fun onCallEnded()
fun onCallError(error: String)
}

七、性能优化与最佳实践

7.1 性能优化策略

/**
* WebRTC性能优化配置
*/
object WebRtcOptimization {

/**
* 优化编码参数
*/
fun optimizeEncoderSettings(sdp: String, isLowBandwidth: Boolean): String {
var modifiedSdp = sdp

if (isLowBandwidth) {
// 降低码率上限
modifiedSdp = modifiedSdp.replace(
"x-google-max-bitrate=\\\\d+".toRegex(),
"x-google-max-bitrate=500"
)

// 降低起始码率
modifiedSdp = modifiedSdp.replace(
"x-google-start-bitrate=\\\\d+".toRegex(),
"x-google-start-bitrate=300"
)
}

return modifiedSdp
}

/**
* 配置低延迟模式
*/
fun configureLowLatency(): MediaConstraints {
return MediaConstraints().apply {
// 禁用音频处理（降低延迟）
optional.add(MediaConstraints.KeyValuePair("googCpuOveruseDetection", "true"))
optional.add(MediaConstraints.KeyValuePair("googPayloadPadding", "false"))

// 优先低延迟
optional.add(MediaConstraints.KeyValuePair("googLatency", "true"))
}
}

/**
* 配置硬件加速
*/
fun enableHardwareAcceleration() {
// 在PeerConnectionFactory构建时配置
// 使用硬件编解码器可降低CPU占用50%+
}
}

7.2 实战效果

通过WebRTC方案，在某智能门铃项目中取得了优异效果：

性能指标：

• 端到端延迟: 60-80ms
• 连接建立时间: 2-3秒
• CPU占用: 15-25%
• 内存占用: 80-120MB
• 流畅度: 98%+ (25fps+)

对比传统方案：

• 延迟降低70% (300ms → 80ms)
• 带宽效率提升40%
• 用户体验提升显著

八、总结

本文系统讲解了WebRTC在Android智能安防系统中的应用实践：

关键要点：

• WebRTC是成熟的实时音视频方案
• PeerConnection是核心API
• 合理配置编解码参数
• 完善的状态管理机制
• 持续监控和优化

参考资料