#!/usr/bin/env python3 """Mock AI Agent — Asterisk/ARI 版. 外呼流程 (控制面纯 REST + RTP; 另加一个原生 socket websocket 仅用于保持 Stasis app 运行, 否则 ARI 会把进入 app 的通道直接挂断): 0. 连接 /ari/events?app=outbound —— 让 Stasis app 处于运行态 (只丢事件, 不解析业务) 1. POST /bridges 建立 mixing 桥 2. POST /channels/externalMedia 外部媒体通道 (ulaw), Asterisk 把桥内混音发到 external_host, 本侧回发地址 = 通道变量 UNICASTRTP_LOCAL_ADDRESS/PORT (兜底: 首包源地址) 3. POST /channels originate 经 PJSIP trunk 外呼被叫 4. 轮询被叫通道至 Up (应答), 双通道入桥 5. RTP 双向: RX=被叫音频(模拟 ASR 输入, RMS 统计); TX=440Hz 间歇音(模拟 LLM/TTS 输出) 6. DELETE 通道+桥 → BYE 纯 stdlib (urllib + asyncio + 原生 socket)。换真实 LLM/ASR: 替换 TX 音源与 RX 消费即可。 用法: mock-agent.py --ari-base http://asterisk1:8088 --external-host <本容器名>:40001 \ --number +15105550123 [--duration 15] """ import argparse, asyncio, base64, json, math, os, random, socket, struct, sys, threading, time import urllib.error, urllib.parse, urllib.request def log(*a): print(f"[agent {time.strftime('%H:%M:%S')}]", *a, flush=True) # ---------- G.711 u-law (与 mock-provider 同源) ---------- def ulaw_encode(s: int) -> int: s = max(-32635, min(32635, s)); sign = 0x80 if s < 0 else 0 if sign: s = -s s += 0x84 e = 7 for i in range(7, -1, -1): if s & (0x40 << i): e = i; break return (~(sign | (e << 4) | ((s >> (e + 3)) & 0x0F))) & 0xFF def ulaw_decode(u: int) -> int: u = ~u & 0xFF sign = u & 0x80; e = (u >> 4) & 0x07; m = u & 0x0F s = ((m << (e + 3)) + 0x84) << 2 return -s if sign else s def tone_payload(ts: float) -> bytes: """20ms / 160 samples @8kHz, 440Hz 500ms-on 500ms-off (模拟 TTS)""" out = bytearray() on = (ts % 1.0) < 0.5 for i in range(160): t = ts + i / 8000.0 v = int(6000 * math.sin(2 * math.pi * 440 * t)) if on else 0 out.append(ulaw_encode(v)) return bytes(out) # ---------- ARI REST ---------- class Ari: def __init__(self, base, user, password): self.base = base.rstrip("/") self.auth = "Basic " + base64.b64encode(f"{user}:{password}".encode()).decode() def req(self, method, path, params=None): url = f"{self.base}/ari{path}" if params: url += "?" + urllib.parse.urlencode(params) r = urllib.request.Request(url, method=method) r.add_header("Authorization", self.auth) try: with urllib.request.urlopen(r, timeout=15) as resp: raw = resp.read() return resp.status, (json.loads(raw) if raw else None) except urllib.error.HTTPError as e: raw = e.read() try: return e.code, (json.loads(raw) if raw else None) except Exception: return e.code, {"error": raw.decode("utf-8", "replace")} # ---------- ARI Stasis app 保活 (极简事件 websocket, 纯 stdlib) ---------- class AppKeeper: """ARI 要求 Stasis app 有活跃 websocket 订阅, 否则进入 app 的通道会被立即挂断. 本类只保持连接并丢弃事件, 控制面仍走 REST.""" def __init__(self, base, app, user, password): u = urllib.parse.urlparse(base) self.host, self.port = u.hostname, u.port or 80 self.app, self.user, self.password = app, user, password self.sock = None def start(self): key = base64.b64encode(os.urandom(16)).decode() req = ("GET /ari/events?app=" + urllib.parse.quote(self.app) + "&api_key=" + urllib.parse.quote(f"{self.user}:{self.password}") + f" HTTP/1.1\r\nHost: {self.host}:{self.port}\r\n" + "Upgrade: websocket\r\nConnection: Upgrade\r\n" + f"Sec-WebSocket-Key: {key}\r\nSec-WebSocket-Version: 13\r\n\r\n") self.sock = socket.create_connection((self.host, self.port), timeout=10) self.sock.sendall(req.encode()) resp = b"" while b"\r\n\r\n" not in resp: chunk = self.sock.recv(1) # 逐字节读响应头, 不吞后续 ws 帧 if not chunk: raise ConnectionError("ws closed during handshake") resp += chunk if b"101" not in resp.split(b"\r\n", 1)[0]: raise ConnectionError(f"ws handshake failed: {resp[:120]!r}") self.sock.settimeout(None) threading.Thread(target=self._drain, daemon=True).start() def _recv_exact(self, n): buf = b"" while len(buf) < n: c = self.sock.recv(n - len(buf)) if not c: raise ConnectionError("ws closed") buf += c return buf def _drain(self): try: while True: b0 = self._recv_exact(1)[0] b1 = self._recv_exact(1)[0] ln = b1 & 0x7F if ln == 126: ln = struct.unpack("!H", self._recv_exact(2))[0] elif ln == 127: ln = struct.unpack("!Q", self._recv_exact(8))[0] payload = self._recv_exact(ln) if ln else b"" op = b0 & 0x0F if op == 0x8: # close break if op == 0x9: # ping → pong mask = os.urandom(4) # client 帧必须带 mask masked = bytes(b ^ mask[i % 4] for i, b in enumerate(payload)) self.sock.sendall(bytes((0x8A, 0x80 | (len(payload) & 0x7F))) + mask + masked) continue if op == 0x1 and payload: try: ev = json.loads(payload) except Exception: continue if ev.get("type") in ("StasisStart", "StasisEnd", "ChannelDestroyed"): log(f"event {ev['type']} {(ev.get('channel') or {}).get('name', '')}") except Exception: pass def stop(self): try: if self.sock: self.sock.close() except Exception: pass # ---------- RTP 媒体 ---------- class Rx(asyncio.DatagramProtocol): def __init__(self, stats): self.stats, self.transport = stats, None def connection_made(self, transport): self.transport = transport def datagram_received(self, data, addr): if len(data) < 12: return self.stats["rx"] += 1 payload = data[12:] dec = [ulaw_decode(b) for b in payload[:40]] self.stats["amp_sum"] += sum(abs(x) for x in dec) / max(1, len(dec)) if self.stats["tx_to"] is None: # 兜底: 用首包源地址作回程 self.stats["tx_to"] = addr if self.stats["rx"] % 250 == 0: log(f"RX frames={self.stats['rx']} avg_rms={self.stats['amp_sum']/self.stats['rx']:.0f}") async def media_loop(listen_port, tx_to, duration): stats = {"rx": 0, "amp_sum": 0.0, "tx": 0, "tx_to": tx_to} loop = asyncio.get_running_loop() rx_t, _ = await loop.create_datagram_endpoint( lambda: Rx(stats), local_addr=("0.0.0.0", listen_port)) async def tx(): seq = random.randint(0, 65535); ts = random.randint(0, 10**6) ssrc = random.randint(1, 2**31) t0 = time.time() while time.time() - t0 < duration: if stats["tx_to"]: payload = tone_payload(time.time() - t0) hdr = struct.pack("!BBHII", 0x80, 0, seq & 0xFFFF, ts & 0xFFFFFFFF, ssrc) rx_t.sendto(hdr + payload, stats["tx_to"]) stats["tx"] += 1 seq += 1; ts += 160 await asyncio.sleep(0.02) try: await asyncio.wait_for(tx(), timeout=duration + 5) finally: rx_t.close() return stats # ---------- 主流程 ---------- def main(): ap = argparse.ArgumentParser() ap.add_argument("--ari-base", default="http://127.0.0.1:8088") ap.add_argument("--ari-user", default="outbound") ap.add_argument("--ari-pass", default="ari_mock_7c3f9e1b5a2d4806") ap.add_argument("--app", default="outbound") ap.add_argument("--external-host", required=True, help="本侧 RTP 地址 ip:port (供 Asterisk 回发混音)") ap.add_argument("--number", default="+15105550123") ap.add_argument("--caller-id", default="+15109990001") ap.add_argument("--trunk", default="mock-trunk") ap.add_argument("--duration", type=float, default=15) ap.add_argument("--ring-timeout", type=float, default=30) args = ap.parse_args() listen_port = int(args.external_host.rsplit(":", 1)[1]) ari = Ari(args.ari_base, args.ari_user, args.ari_pass) suffix = f"{int(time.time())}-{random.randint(100, 999)}" bridge_id, em_id, callee_id = f"br-{suffix}", f"em-{suffix}", f"callee-{suffix}" # 0. 保持 Stasis app 运行 (否则通道进 app 即被挂断) keeper = AppKeeper(args.ari_base, args.app, args.ari_user, args.ari_pass) keeper.start() log(f"stasis app '{args.app}' running (event ws connected)") def fail(msg, detail=None): log(f"FAIL: {msg}", detail or "") cleanup() sys.exit(1) def cleanup(): for m, p in (("DELETE", f"/channels/{callee_id}"), ("DELETE", f"/channels/{em_id}"), ("DELETE", f"/bridges/{bridge_id}")): try: ari.req(m, p) except Exception: pass keeper.stop() # 1. 混音桥 st, r = ari.req("POST", "/bridges", params={"type": "mixing", "bridgeId": bridge_id}) if st not in (200, 201): fail("create bridge", (st, r)) # 2. 外部媒体通道 (Asterisk → 本侧 RTP) st, em = ari.req("POST", "/channels/externalMedia", params={ "app": args.app, "external_host": args.external_host, "format": "ulaw", "channelId": em_id}) if st not in (200, 201): fail("create externalMedia", (st, r)) log(f"externalMedia channel up: {em.get('name')}") # 3. Asterisk 侧 RTP 回送地址 tx_to = None st, v = ari.req("GET", f"/channels/{em_id}/variable", {"variable": "UNICASTRTP_LOCAL_ADDRESS"}) ip = (v or {}).get("value") if st == 200 else None st, v = ari.req("GET", f"/channels/{em_id}/variable", {"variable": "UNICASTRTP_LOCAL_PORT"}) port = (v or {}).get("value") if st == 200 else None if ip and port: tx_to = (ip, int(port)) log(f"asterisk RTP return addr: {ip}:{port}") else: log("UNICASTRTP vars not set; will reply-to-source on first RX") # 4. 发起外呼 st, ch = ari.req("POST", "/channels", params={ "endpoint": f"PJSIP/{args.number}@{args.trunk}", "app": args.app, "appArgs": "outbound", "callerId": args.caller_id, "channelId": callee_id, "timeout": int(args.ring_timeout)}) if st not in (200, 201): fail("originate", (st, ch)) log(f"INVITE sent via {args.trunk} -> {args.number} (channel {callee_id})") # 5. 等待被叫应答 deadline = time.time() + args.ring_timeout state = None while time.time() < deadline: st, ch = ari.req("GET", f"/channels/{callee_id}") if st == 200: state = ch.get("state") elif st in (404, 410): fail("callee channel gone before answer") if state == "Up": break time.sleep(0.25) if state != "Up": fail(f"no answer (state={state})") log(f"callee answered: state={state}") # 6. 双通道入桥 → 双向 RTP st, r = ari.req("POST", f"/bridges/{bridge_id}/addChannel", params={"channel": f"{em_id},{callee_id}"}) if st not in (200, 201, 204): fail("addChannel", (st, r)) log(f"bridged [{em_id} + {callee_id}], streaming {args.duration}s " f"(RX=ASR mock, TX=440Hz LLM/TTS mock)") # 7. 媒体收发 stats = asyncio.run(media_loop(listen_port, tx_to, args.duration)) # 8. 挂断收尾 cleanup() n = stats["rx"] amp = stats["amp_sum"] / n if n else 0.0 print(f"RESULT number={args.number} rx_frames={n} rx_avg_rms={amp:.0f} " f"tx_frames={stats['tx']} bidirectional={'YES' if n > 50 and amp > 100 else 'NO'}", flush=True) if __name__ == "__main__": main()