使用aiortc库，设置音频传输的源和目标，建立简单但无双工音频通信服务器。

代码示例：

import asyncio
import cv2
import numpy as np
from aiortc import AudioStreamTrack, RTCPeerConnection, RTCSessionDescription
from aiortc.contrib.media import MediaPlayer, MediaRecorder

# Audio stream track to handle audio data
class AudioStream(AudioStreamTrack):
    def __init__(self):
        super().__init__()
        self.recorder = MediaRecorder("audio.wav")

    async def recv(self):
        # Wait for data and return the recorded audio
        return await self.recorder.recv()

# Signaling and WebRTC connections
async def handle_client(reader, writer):
    # Get the offer from the client
    data = await reader.read()
    print("Received offer from client")
    pc = RTCPeerConnection()
    await pc.setRemoteDescription(RTCSessionDescription(sdp=data, type="offer"))

    # Create the audio stream and add to the peer connection
    audio_track = AudioStream()
    pc.addTrack(audio_track)

    # Generate answer and send back to client
    await pc.setLocalDescription(await pc.createAnswer())
    response = {"sdp": pc.localDescription.sdp, "type": pc.localDescription.type}
    writer.write(str.encode(json.dumps(response)))
    await writer.drain()
    print("Sent answer to client")

async def server():
    server = await asyncio.start_server(handle_client, port=1234)
    print("Server started...")
    async with server:
        await server.serve_forever()

if __name__ == "__main__":
    asyncio.run(server())

该示例可以通过以下方法进行测试：

1. 安装 aiortc 和 opencv-python 库
2. 运行示例代码
3. 使用WebRTC发送音频流到服务器
4. 服务器将自动接收并记录音频流