Files
ai-video-fullstack/backend/routes/voice_webrtc.py
Xin Wang ede5e2334b Implement WebRTC offer handling and ICE candidate management in voice_webrtc.py
- Add new HTTP endpoints for handling WebRTC offers and ICE candidates, enhancing the signaling process for voice interactions.
- Introduce dynamic variable decoding from request headers to support flexible offer payloads.
- Refactor existing WebSocket handling to accommodate new offer processing logic.
- Update frontend dependencies to include Pipecat client libraries for improved WebRTC transport management.
- Streamline voice preview functionality by integrating SmallWebRTCTransport for better media handling.
2026-07-13 06:50:23 +08:00

227 lines
8.4 KiB
Python

"""WebRTC 输出:SmallWebRTC 信令握手。
参考 dograh 的 webrtc_signaling.py,砍掉鉴权/配额/DB/org/ICE 过滤策略/TURN。
握手消息:
client → {type:"offer", payload:{pc_id, sdp, type, assistant_id}}
server → {type:"answer", payload:{pc_id, sdp, type}}
both → {type:"ice-candidate", payload:{pc_id, candidate:{...}}}
server → {type:"error", payload:{message}}
"""
import asyncio
import base64
import json
from db.session import SessionLocal
from fastapi import APIRouter, Body, Depends, Request, WebSocket
from loguru import logger
from models import AssistantConfig, SignalingOffer
from services.auth import require_admin, require_admin_websocket
from services.config_resolver import resolve_runtime_config
from services.runtime_variables import DynamicVariableError, prepare_dynamic_config
from starlette.websockets import WebSocketDisconnect, WebSocketState
from services.webrtc_ice import aiortc_ice_servers, client_ice_servers
# 注意:pipecat 是重依赖(语音才用),在 _handle_offer 等处惰性导入。
router = APIRouter(tags=["voice"])
_http_peers: dict[str, object] = {}
@router.get("/api/webrtc/ice-servers", dependencies=[Depends(require_admin)])
async def ice_servers():
"""Browser fetches STUN/TURN config (with ephemeral TURN creds when configured)."""
return {"iceServers": client_ice_servers()}
@router.post("/api/webrtc/offer", dependencies=[Depends(require_admin)])
async def http_offer(request: Request, payload: dict = Body(...)):
"""Official SmallWebRTC JS transport offer endpoint."""
request_data = payload.get("requestData") or {}
encoded_variables = request.headers.get("x-pipecat-dynamic-variables", "")
header_variables = {}
if encoded_variables:
padding = "=" * (-len(encoded_variables) % 4)
header_variables = json.loads(
base64.urlsafe_b64decode(encoded_variables + padding).decode("utf-8")
)
offer_payload = {
**payload,
"assistant_id": request.headers.get("x-pipecat-assistant-id"),
"dynamic_variables": header_variables,
**request_data,
}
answer = await _handle_offer_payload(offer_payload, _http_peers)
return answer
@router.patch("/api/webrtc/offer", dependencies=[Depends(require_admin)])
async def http_ice_candidates(payload: dict = Body(...)):
"""Accept the batched trickle ICE format used by the official JS client."""
pc_id = payload.get("pc_id")
pc = _http_peers.get(pc_id) if pc_id else None
if not pc:
return {"ok": False}
from aiortc.sdp import candidate_from_sdp
for item in payload.get("candidates") or []:
candidate = candidate_from_sdp(item.get("candidate", ""))
candidate.sdpMid = item.get("sdp_mid")
candidate.sdpMLineIndex = item.get("sdp_mline_index")
await pc.add_ice_candidate(candidate)
return {"ok": True}
@router.websocket("/ws/voice")
async def voice_signaling(websocket: WebSocket):
if not await require_admin_websocket(websocket):
return
await websocket.accept()
peers: dict = {}
try:
while True:
message = await websocket.receive_json()
try:
if message.get("type") == "offer":
await _handle_offer(websocket, message.get("payload", {}), peers)
elif message.get("type") == "ice-candidate":
await _handle_ice(message.get("payload", {}), peers)
except Exception as e:
logger.exception(f"处理 WebRTC 信令消息失败: {e}")
if websocket.application_state == WebSocketState.CONNECTED:
await websocket.send_json(
{
"type": "error",
"payload": {
"message": (
str(e)
if isinstance(e, DynamicVariableError)
else f"语音会话启动失败: {type(e).__name__}"
)
},
}
)
except WebSocketDisconnect:
logger.info("WebRTC 信令断开")
except Exception as e:
logger.error(f"WebRTC 信令出错: {e}")
finally:
# disconnect() triggers the registered closed callback, which removes
# the peer from this dict. Iterate over a snapshot to avoid mutation.
for pc in list(peers.values()):
await pc.disconnect()
async def _resolve_config(offer: SignalingOffer) -> AssistantConfig:
"""优先用 assistant_id 从 DB 解析(含真 key);否则用调试内联配置。"""
if offer.assistant_id:
async with SessionLocal() as session:
cfg = await resolve_runtime_config(session, offer.assistant_id)
return prepare_dynamic_config(
cfg,
offer.dynamic_variables,
assistant_id=offer.assistant_id,
)
if offer.inline_config:
return prepare_dynamic_config(
offer.inline_config,
offer.dynamic_variables,
assistant_id=None,
)
raise ValueError("offer 缺少 assistant_id 或 inline_config")
async def _handle_offer(websocket, payload, peers):
answer = await _handle_offer_payload(payload, peers)
if websocket.application_state == WebSocketState.CONNECTED:
await websocket.send_json(
{
"type": "answer",
"payload": answer,
}
)
async def _handle_offer_payload(payload, peers):
from pipecat.transports.smallwebrtc.connection import SmallWebRTCConnection
from services.pipecat.pipeline import run_pipeline
from services.pipecat.transports import build_webrtc_transport
offer = SignalingOffer(**payload)
pc_id = offer.pc_id
restart_pc = bool(payload.get("restart_pc"))
if restart_pc and pc_id and pc_id in peers:
old_pc = peers.pop(pc_id)
await old_pc.disconnect()
if pc_id and pc_id in peers:
pc = peers[pc_id]
await pc.renegotiate(sdp=offer.sdp, type=offer.type, restart_pc=False)
else:
cfg = await _resolve_config(offer) # 解析放在建连前,配置错就别建连
# 服务端助手配置是视觉理解的唯一授权来源;客户端 offer 只负责携带媒体轨。
vision_enabled = cfg.vision_enabled
if vision_enabled:
has_native_vision = (
not cfg.vision_model_resource_id and cfg.llm_support_image_input
)
has_aux_vision_model = (
bool(cfg.vision_model_resource_id)
and cfg.vision_llm_support_image_input
)
if not (has_native_vision or has_aux_vision_model):
raise ValueError(
"当前模型不支持图片输入,请在模型资源中选择支持图片输入的视觉模型"
)
pc = SmallWebRTCConnection(ice_servers=aiortc_ice_servers())
if pc_id:
pc._pc_id = pc_id
await pc.initialize(sdp=offer.sdp, type=offer.type)
peers[pc.pc_id] = pc
@pc.event_handler("closed")
async def _on_closed(conn: SmallWebRTCConnection):
peers.pop(conn.pc_id, None)
# 后台跑管线:WebRTC transport + 解析出的运行时配置
transport = build_webrtc_transport(
pc,
video_in_enabled=vision_enabled,
)
asyncio.create_task(
run_pipeline(
transport,
cfg,
vision_enabled=vision_enabled,
assistant_id=offer.assistant_id,
channel="webrtc",
)
)
answer = pc.get_answer()
return {
"pc_id": answer["pc_id"],
"sdp": answer["sdp"],
"type": answer["type"],
}
async def _handle_ice(payload, peers):
from aiortc.sdp import candidate_from_sdp
pc_id = payload.get("pc_id")
candidate_data = payload.get("candidate")
pc = peers.get(pc_id) if pc_id else None
if not pc or not candidate_data:
return
try:
candidate = candidate_from_sdp(candidate_data.get("candidate", ""))
candidate.sdpMid = candidate_data.get("sdpMid")
candidate.sdpMLineIndex = candidate_data.get("sdpMLineIndex")
await pc.add_ice_candidate(candidate)
except Exception as e:
logger.error(f"添加 ICE candidate 失败: {e}")