Add WebRTC TURN configuration and related enhancements
- Introduce a new `.env.example` file for environment variable setup, including `PUBLIC_IP`, `TURN_SECRET`, and `TURN_URLS` for WebRTC TURN server configuration. - Update `docker-compose.yaml` to support TURN server deployment with necessary environment variables and commands. - Enhance backend configuration and routes to include WebRTC ICE server settings, allowing for STUN/TURN server integration. - Implement a new service for managing WebRTC ICE server configurations, providing credentials for TURN when configured. - Modify frontend API to fetch ICE server configurations dynamically, improving support for cross-network voice preview.
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4
.env.example
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4
.env.example
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# docker compose 变量(复制为项目根 .env)。公网 WebRTC 需配合 --profile remote。
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PUBLIC_IP=182.92.86.220
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TURN_SECRET=change-me-to-a-long-random-string
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TURN_URLS=turn:182.92.86.220:3478?transport=udp,turn:182.92.86.220:3478?transport=tcp
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@@ -31,5 +31,12 @@ DATABASE_URL=postgresql+asyncpg://postgres:postgres@localhost:5432/postgres
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# ---- 服务监听 & 跨域 ----
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HOST=0.0.0.0
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PORT=8000
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# 前端开发地址,允许跨域
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# 前端开发地址,允许跨域(公网部署时加上实际前端 origin)
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CORS_ORIGINS=http://localhost:3000,http://127.0.0.1:3000
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# ---- WebRTC TURN(公网跨网语音预览;本地开发留空) ----
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# 与 docker compose --profile remote 的 coturn 配套。云安全组需放行 UDP 3478 与 49152-49200。
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# PUBLIC_IP 填云主机公网 IP(compose 里给 coturn --external-ip 用,见项目根 .env)。
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# TURN_URLS=turn:182.92.86.220:3478?transport=udp,turn:182.92.86.220:3478?transport=tcp
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# TURN_SECRET=your-turn-secret
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# TURN_CREDENTIAL_TTL=86400
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@@ -9,6 +9,7 @@
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/api/model-resources 模型资源 CRUD
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/ws/voice WebRTC 输出(浏览器)
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/ws/stream WS 输出(裸音频流)
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/api/webrtc/ice-servers WebRTC STUN/TURN 配置
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"""
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from contextlib import asynccontextmanager
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@@ -43,3 +43,13 @@ PORT = int(os.getenv("PORT", "8000"))
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CORS_ORIGINS = _split(
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os.getenv("CORS_ORIGINS", "http://localhost:3000,http://127.0.0.1:3000")
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)
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# ---- WebRTC TURN(公网跨网语音预览;本地开发留空即可) ----
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# TURN_URLS 示例:turn:182.92.86.220:3478?transport=udp,turn:182.92.86.220:3478?transport=tcp
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TURN_URLS = _split(os.getenv("TURN_URLS", ""))
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# coturn --use-auth-secret 时与 --static-auth-secret 一致;后端据此签发短时凭证
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TURN_SECRET = os.getenv("TURN_SECRET", "")
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# 不用 secret 时可改静态账号(需 coturn --user=...)
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TURN_USERNAME = os.getenv("TURN_USERNAME", "")
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TURN_PASSWORD = os.getenv("TURN_PASSWORD", "")
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TURN_CREDENTIAL_TTL = int(os.getenv("TURN_CREDENTIAL_TTL", "86400"))
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@@ -17,17 +17,17 @@ from models import AssistantConfig, SignalingOffer
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from services.config_resolver import resolve_runtime_config
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from starlette.websockets import WebSocketDisconnect, WebSocketState
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# 注意:pipecat / aiortc 都是重依赖(语音才用),改成函数内"惰性导入",
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# 这样不装 pipecat 也能启动后端、验证 CRUD。语音真正用到时才加载。
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from services.webrtc_ice import aiortc_ice_servers, client_ice_servers
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router = APIRouter()
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# 注意:pipecat 是重依赖(语音才用),在 _handle_offer 等处惰性导入。
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router = APIRouter(tags=["voice"])
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def _ice_servers():
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from aiortc import RTCIceServer
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# 本地只用 STUN;公网部署在此追加 TURN(参考 dograh get_ice_servers)
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return [RTCIceServer(urls="stun:stun.l.google.com:19302")]
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@router.get("/api/webrtc/ice-servers")
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async def ice_servers():
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"""Browser fetches STUN/TURN config (with ephemeral TURN creds when configured)."""
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return {"iceServers": client_ice_servers()}
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@router.websocket("/ws/voice")
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@@ -87,7 +87,7 @@ async def _handle_offer(websocket, payload, peers):
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await pc.renegotiate(sdp=offer.sdp, type=offer.type, restart_pc=False)
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else:
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cfg = await _resolve_config(offer) # 解析放在建连前,配置错就别建连
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pc = SmallWebRTCConnection(ice_servers=_ice_servers())
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pc = SmallWebRTCConnection(ice_servers=aiortc_ice_servers())
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if pc_id:
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pc._pc_id = pc_id
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await pc.initialize(sdp=offer.sdp, type=offer.type)
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60
backend/services/webrtc_ice.py
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backend/services/webrtc_ice.py
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"""WebRTC ICE server config: STUN + optional TURN for cross-network voice preview."""
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from __future__ import annotations
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import base64
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import hashlib
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import hmac
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import time
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import config
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STUN_URL = "stun:stun.l.google.com:19302"
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def _turn_credentials() -> tuple[str, str] | None:
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"""Return (username, credential) for TURN, or None when TURN is not configured."""
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if not config.TURN_URLS:
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return None
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if config.TURN_SECRET:
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expiry = int(time.time()) + config.TURN_CREDENTIAL_TTL
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username = f"{expiry}:ai-video"
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credential = base64.b64encode(
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hmac.new(
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config.TURN_SECRET.encode("utf-8"),
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username.encode("utf-8"),
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hashlib.sha1,
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).digest()
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).decode("utf-8")
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return username, credential
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if config.TURN_USERNAME and config.TURN_PASSWORD:
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return config.TURN_USERNAME, config.TURN_PASSWORD
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return None
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def client_ice_servers() -> list[dict]:
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"""ICE servers for browser RTCPeerConnection (JSON-serializable)."""
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servers: list[dict] = [{"urls": STUN_URL}]
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creds = _turn_credentials()
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if not creds:
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return servers
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username, credential = creds
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for url in config.TURN_URLS:
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servers.append({"urls": url, "username": username, "credential": credential})
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return servers
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def aiortc_ice_servers() -> list:
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"""ICE servers for backend SmallWebRTCConnection / aiortc."""
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from aiortc import RTCIceServer
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servers = [RTCIceServer(urls=STUN_URL)]
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creds = _turn_credentials()
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if not creds:
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return servers
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username, credential = creds
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for url in config.TURN_URLS:
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servers.append(
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RTCIceServer(urls=url, username=username, credential=credential)
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)
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return servers
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@@ -49,6 +49,9 @@ services:
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DATABASE_URL: "postgresql+asyncpg://postgres:${POSTGRES_PASSWORD:-postgres}@postgres:5432/postgres"
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# 3030 = docker ui 宿主端口;3000 = 宿主机裸跑 npm run dev 时的端口
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CORS_ORIGINS: "http://localhost:3030,http://127.0.0.1:3030,http://localhost:3000,http://127.0.0.1:3000"
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# WebRTC TURN(公网部署:设 PUBLIC_IP + TURN_SECRET,并 docker compose --profile remote up)
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TURN_URLS: "${TURN_URLS:-}"
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TURN_SECRET: "${TURN_SECRET:-}"
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ports:
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- "8000:8000"
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depends_on:
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@@ -112,17 +115,24 @@ services:
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networks: [app-network]
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# ---- 可选(profile: remote):WebRTC 公网穿透 ----
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# 在项目根 .env 设置 PUBLIC_IP(云主机公网 IP)与 TURN_SECRET,与 backend TURN_SECRET 一致。
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# 云安全组放行:UDP/TCP 3478,UDP 49152-49200。
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coturn:
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image: coturn/coturn:4.8.0
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profiles: ["remote"]
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network_mode: host # TURN 需直接占用 UDP 端口段
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network_mode: host
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command:
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- -n
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- --log-file=stdout
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- --listening-port=3478
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- --listening-ip=0.0.0.0
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- --external-ip=${PUBLIC_IP:?set PUBLIC_IP in .env for coturn}
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- --realm=ai-video
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- --use-auth-secret
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- --static-auth-secret=${TURN_SECRET:-changeme}
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- --min-port=49152
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- --max-port=49200
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- --no-cli
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# ---- 可选(profile: tls):nginx 反代统一 TLS,局域网 https 调试语音预览 ----
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# 起前先生成证书:./deploy/setup-certs.sh(证书落在 deploy/certs/)
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@@ -19,7 +19,7 @@
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import { useCallback, useEffect, useRef, useState } from "react";
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import { API_BASE } from "@/lib/api";
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import { API_BASE, webrtcApi } from "@/lib/api";
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export type VoicePreviewStatus = "idle" | "connecting" | "connected" | "failed";
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@@ -301,10 +301,12 @@ export function useVoicePreview(
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if (wsRef.current === ws) closeOnRemoteEnd("语音信令连接已断开。");
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};
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// 2) 建 PeerConnection(纯 STUN,本机/局域网够用)
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const pc = new RTCPeerConnection({
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iceServers: [{ urls: "stun:stun.l.google.com:19302" }],
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});
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// 2) 建 PeerConnection(STUN;公网跨网时后端会下发 TURN)
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const iceServers = await webrtcApi
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.iceServers()
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.then((r) => r.iceServers)
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.catch(() => [{ urls: "stun:stun.l.google.com:19302" }]);
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const pc = new RTCPeerConnection({ iceServers });
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pcRef.current = pc;
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pc.onicecandidate = (e) => {
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@@ -208,3 +208,14 @@ export type NodeTypesResponse = {
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export const nodeTypesApi = {
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list: () => request<NodeTypesResponse>("/api/node-types"),
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};
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export type IceServerConfig = {
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urls: string | string[];
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username?: string;
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credential?: string;
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};
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export const webrtcApi = {
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iceServers: () =>
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request<{ iceServers: IceServerConfig[] }>("/api/webrtc/ice-servers"),
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};
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