WebRTC安全与媒体权限治理最佳实践

WebRTC安全与媒体权限治理最佳实践概述WebRTC提供实时媒体访问与传输能力。通过严格的权限与设备选择、网络策略与加密传输，可降低隐私与安全风险。权限与设备选择async function requestMedia(constraints: MediaStreamConstraints): Promise<MediaStream | null> {

try {

return await navigator.mediaDevices.getUserMedia(constraints)

} catch (e) {

return null

}

}

async function listDevices(): Promise<MediaDeviceInfo[]> {

return await navigator.mediaDevices.enumerateDevices()

}

const audioOnly: MediaStreamConstraints = { audio: { echoCancellation: true, noiseSuppression: true }, video: false }

网络策略const rtcConfig: RTCConfiguration = {

iceServers: [

{ urls: ['stun:stun.l.google.com:19302'] },

{ urls: ['turn:turn.example.com:3478'], username: 'user', credential: 'pass' }

],

iceTransportPolicy: 'all'

}

function createPeer(): RTCPeerConnection {

const pc = new RTCPeerConnection(rtcConfig)

pc.onicecandidate = (e) => { /* 发送到信令服务器 */ }

return pc

}

加密传输与会话治理function attachStream(pc: RTCPeerConnection, stream: MediaStream) {

for (const track of stream.getTracks()) pc.addTrack(track, stream)

}

function closeSession(pc: RTCPeerConnection) {

pc.getSenders().forEach(s => s.track && s.track.stop())

pc.close()

}

运维要点仅在HTTPS环境下使用WebRTC并限制设备访问范围TURN服务器使用安全凭证与最小暴露网络策略对会话建立与关闭进行审计并限制时长与并发通过权限、设备与网络策略治理，结合加密传输，可实现安全可控的WebRTC媒体访问与传输。