第三方媒体流压力测试:k6插件xk6-webrtc的使用来测试媒体流的性能
1. 安装构建
首先需要构建包含webrtc插件的k6二进制文件：
# 安装 xk6
go install go.k6.io/xk6/cmd/xk6@latest# 构建包含 webrtc 插件的 k6
xk6 build --with github.com/grafana/xk6-webrtc@latest# 或者指定版本
xk6 build --with github.com/grafana/xk6-webrtc@v0.3.0构建完成后会生成一个k6可执行文件。
2. 基础使用
一个基本的WebRTC测试脚本：
// basic-webrtc-test.js
import { WebRTC } from 'k6/experimental/webrtc';
import { check } from 'k6';// 信令服务器地址
const SIGNALING_SERVER = 'ws://localhost:8080/signaling';export default function () {// 创建 WebRTC 配置const client = new WebRTC({iceServers: [{ urls: ['stun:stun.l.google.com:19302'] },// 如果需要 TURN 服务器// { urls: 'turn:your-turn-server.com', username: 'user', credential: 'pass' }],});// 创建模拟媒体流const stream = client.createStream({ audio: true, video: true });client.addStream(stream);// 设置事件处理器client.onconnectionstatechange = (state) => {console.log(`Connection state changed to: ${state}`);};client.oniceconnectionstatechange = (state) => {console.log(`ICE connection state changed to: ${state}`);};// 连接到信令服务器const ws = new WebSocket(SIGNALING_SERVER);ws.onopen = () => {console.log('Connected to signaling server');// 创建 offerclient.createOffer().then((offer) => {// 设置本地描述return client.setLocalDescription(offer);}).then(() => {// 通过信令服务器发送 offerws.send(JSON.stringify({type: 'offer',sdp: client.localDescription}));}).catch((err) => {console.error('Error creating offer:', err);});};// 处理来自信令服务器的消息ws.onmessage = (event) => {const message = JSON.parse(event.data);if (message.type === 'answer') {// 收到 answer，设置远程描述client.setRemoteDescription(message.sdp).then(() => {console.log('Remote description set successfully');}).catch((err) => {console.error('Error setting remote description:', err);});} else if (message.candidate) {// 添加 ICE candidateclient.addIceCandidate(message.candidate).then(() => {console.log('ICE candidate added');}).catch((err) => {console.error('Error adding ICE candidate:', err);});}};// 检查连接状态check(client, {'connection established': (c) => c.connectionState === 'connected','ice connection successful': (c) => c.iceConnectionState === 'connected',});// 收集统计信息const stats = client.getStats();for (let stat of stats) {console.log(`Stat: ${stat.type} - ${stat.id}: ${stat.value}`);}
}export function setup() {// 测试设置，可选console.log('Starting WebRTC load test');
}export function teardown() {// 测试清理，可选console.log('WebRTC load test completed');
}3. 高级功能
数据通道测试
// data-channel-test.js
import { WebRTC } from 'k6/experimental/webrtc';export default function () {const client = new WebRTC({iceServers: [{ urls: ['stun:stun.l.google.com:19302'] }]});// 创建数据通道const dataChannel = client.createDataChannel('test-channel', {ordered: true,maxRetransmits: 3});dataChannel.onopen = () => {console.log('Data channel opened');// 发送测试数据for (let i = 0; i < 10; i++) {dataChannel.send(`Test message ${i}`);}};dataChannel.onmessage = (event) => {console.log(`Received message: ${event.data}`);};dataChannel.onclose = () => {console.log('Data channel closed');};// 其余信令逻辑与基础示例相同...
}媒体流统计监控
// media-stats-test.js
import { WebRTC } from 'k6/experimental/webrtc';
import { Trend, Rate, Counter } from 'k6/metrics';// 自定义指标
const audioBitrate = new Trend('webrtc_audio_bitrate_bps');
const videoBitrate = new Trend('webrtc_video_bitrate_bps');
const packetLoss = new Rate('webrtc_packet_loss_rate');
const connectionSuccess = new Counter('webrtc_connection_success');export default function () {const client = new WebRTC({iceServers: [{ urls: ['stun:stun.l.google.com:19302'] }]});// 创建媒体流const stream = client.createStream({ audio: { bitrate: 64000, // 64 kbpscodec: 'OPUS'}, video: { bitrate: 500000, // 500 kbpscodec: 'VP8'} });client.addStream(stream);// 定期收集统计信息const interval = setInterval(() => {const stats = client.getStats();stats.forEach(stat => {switch(stat.type) {case 'outbound-rtp':if (stat.kind === 'audio') {audioBitrate.add(stat.bitrate || 0);} else if (stat.kind === 'video') {videoBitrate.add(stat.bitrate || 0);}break;case 'candidate-pair':if (stat.state === 'succeeded') {connectionSuccess.add(1);}break;}});}, 1000); // 每秒收集一次// 测试完成后清理setTimeout(() => {clearInterval(interval);client.close();}, 30000); // 运行30秒
}
4. 运行测试
# 运行基础测试
./k6 run basic-webrtc-test.js# 带虚拟用户的负载测试
./k6 run --vus 10 --duration 30s basic-webrtc-test.js# 分段测试
./k6 run --stages 30s:10,1m:20,30s:10 basic-webrtc-test.js# 输出详细结果
./k6 run --summary-export=results.json basic-webrtc-test.js5. 配置选项
WebRTC配置
const client = new WebRTC({// ICE 服务器配置iceServers: [{ urls: 'stun:stun.zmtests.com:19302' },{ urls: 'turn:turn.zmtests.com:3478',username: 'user',credential: 'password'}],// ICE 传输策略iceTransportPolicy: 'all', // 'relay' | 'all'// Bundle 策略bundlePolicy: 'balanced', // 'balanced' | 'max-compat' | 'max-bundle'// RTCP mux 策略rtcpMuxPolicy: 'require', // 'require' | 'negotiate'// 证书certificates: [] // 可选的证书数组
});媒体流配置
const stream = client.createStream({audio: {bitrate: 128000, // 比特率 (bps)codec: 'OPUS', // 编码器sampleRate: 48000, // 采样率channelCount: 2 // 声道数},video: {bitrate: 1000000, // 比特率 (bps)codec: 'VP8', // 编码器width: 1280, // 宽度height: 720, // 高度frameRate: 30 // 帧率}
});6. 注意事项
信令服务器：需要单独实现信令服务器来处理SDP和ICE candidate交换
媒体服务器：需要真实的媒体服务器（如 Mediasoup, Jitsi, Pion）来处理媒体流
资源限制：大量WebRTC连接会消耗大量CPU和网络资源
网络模拟：可以使用 xk6-net 插件模拟网络条件
错误处理：务必添加适当的错误处理，因为WebRTC连接可能因为各种原因失败
这个插件为WebRTC应用的负载测试提供了强大的测试能力，特别适合测试信令服务器和基础连接性能。