diff --git a/freeswitch/README.md b/freeswitch/README.md new file mode 100644 index 0000000..130df18 --- /dev/null +++ b/freeswitch/README.md @@ -0,0 +1,177 @@ +# FreeSWITCH/ESL AI 外呼底座 — 部署与运维文档 + +> 目标机器: `39.106.106.246` (阿里云 2vCPU/3.5G, Debian 13) · 交付物: `/opt/freeswitch-esl/` +> 项目: https://github.com/signalwire/freeswitch · 镜像: safarov/freeswitch (FreeSWITCH 1.10.12) +> 范围: **仅外呼** (ESL originate ×2 + uuid_bridge), Mock SIP 运营商 + Mock LLM/ASR, 已端到端验证双向语音。 +> 前置: asterisk 运行时已下线 (ast-* 容器清零、8088/8089 释放); Docker 与内核调优沿用 livekit 期成果。 + +## 1. 水平扩展结论 (先行) + +**结论: 零共享状态的独立节点复制 —— 与 Asterisk 同构。FreeSWITCH 无内置集群, 每节点呼叫状态全在本机内存; 扩容 = 加一个节点, 呼叫发起侧 (agent/业务后端) 在节点名单上选节点建 ESL 连接。不需要 Redis, 不需要任何协调组件。** + +| 层 | 状态存放 | 扩容方式 | 调度机制 | +|---|---|---|---| +| FreeSWITCH 节点 | 本机内存 (通道/桥) | 复制 service 块 (独立宿主 ESL 端口) | 呼叫发起侧选节点 (call.sh 的 NODE 参数 = 生产中的 LB 名单) | +| AI Agent (生产) | 无状态 worker | `--scale`/K8s HPA | 每个外呼任务自带节点选择与 ESL/SIP/RTP 会话 | + +三方案对比: livekit-sip 需 Redis 存 trunk/呼叫状态并派发; Asterisk (ARI) 与 FreeSWITCH (ESL) 出站外呼均零共享层, 每呼叫自包含 (originate ×2 + bridge 全在一个节点内闭环), 节点间零交互。实测 1→2 节点**零配置变更** (共用同一套 etc-fs), 呼叫按发起侧选择分摊 (见 §7)。 + +瓶颈与边界: ① 单节点容量受 CPU 与 RTP 端口段限制 — vanilla 默认 16384-32768 (约 1.6 万端口, 每桥接呼叫 2 腿各 1 对 RTP+RTCP ≈ 8000 并发/节点, 容器 fd/CPU 先到顶); ② ESL 是单连接同步命令式接口, 高并发控制面需连接池或 bgapi; ③ 无跨节点媒体互通 — 仅外呼场景无影响; ④ 未来要呼入/注册时前置 Kamailio dispatcher, 纯出站不需要。 + +## 2. 架构 + +``` +业务后端 / AI Agent (mock-agent, 容器, 纯 stdlib Python) + │ ① ESL TCP 连接 fs:8021 (auth) — 控制面 (等价 ARI REST) + │ ② 本侧 SIP UAS 监听 :5062 + RTP :40001 — 媒体面 (等价 ARI externalMedia) + │ ③ originate 腿1 → sofia/external/agent@me:5062 &park() + │ ④ originate 腿2 → sofia/gateway/mock-trunk/<被叫> &park() + │ ⑤ uuid_bridge 腿1 腿2 → RTP 经 FreeSWITCH 双向 + ▼ +┌─ docker net: fsnet ─────────────────────────────────────────────┐ +│ fs1 (ESL :8021, sofia ext :5080/udp, RTP 16384+) ← 独立节点 │ +│ fs2 (ESL :8021→宿主 8022, 其余同上) ← 零共享状态 │ +│ │ INVITE / 100/180 / 200 OK (SDP, PCMU) / ACK │ +│ ▼ │ +│ mock-provider (SIP UAS :5060/udp, RTP :40000, 440Hz 应答音) │ +└──────────────────────────────────────────────────────────────────┘ + agent ⇄ FreeSWITCH: RTP 双向 (PCMU) + TX = 440Hz 间歇音 (Mock LLM/TTS) → 被叫听到的 "AI 说话" + RX = RMS 统计 (Mock ASR) → 被叫语音进 "识别" +``` + +**与 Asterisk 方案的关键差异**: FreeSWITCH **没有 ARI externalMedia 式外挂媒体 API**。标准做法是 agent 自身充当一个 SIP 端点 (UAS): FreeSWITCH originate 一条到 agent 的 SIP 腿 (agent 应答 200 OK+SDP), 再 originate 被叫腿, `uuid_bridge` 桥接。控制面从 "ARI REST + 事件 WS" 变为 "ESL TCP 单连接命令式"。agent 的 RTP 收发逻辑与 asterisk 版相同, 仅增加 SIP UAS 部分。 + +外呼信令流 (仅出站): +``` +agent → ESL auth → originate sofia/external/agent@host:5062 &park() (腿1, agent 应答) + → originate sofia/gateway/mock-trunk/+1510... &park() (腿2, 被叫应答) + → uuid_bridge <腿1> <腿2> → 双向 RTP 经 FS + → uuid_kill 双腿 → BYE → agent 输出双向统计 +``` + +## 3. 端口规划 + +| 端口 | 组件 | 说明 | +|---|---|---| +| 8021/tcp | fs1 | ESL 控制面, 宿主映射 8021 | +| 8022/tcp→8021 | fs2 | 第二节点 ESL (profile `scale`) | +| 5080/udp | fs1/fs2 | sofia external profile 信令 (容器网内, 未映射宿主) | +| 16384-32768/udp | fs1/fs2 | RTP 媒体 (vanilla 默认段, 各容器独立 netns 不冲突) | +| 5060/udp, 40000/udp | mock-provider | Mock 运营商 SIP + RTP | +| 5062/udp, 40001/udp | mock-agent (每次呼叫) | 本侧 SIP UAS + RTP | + +生产外网仅需放行: SIP 5060/udp (+5061/tls) 与 RTP 段; **ESL 8021/8022 只对 agent 内网放行 (强密码+ACL), 严禁公网裸开** — ESL 拿到即完全控制 FreeSWITCH (可执行任意系统命令 `system`)。 + +## 4. 从零部署步骤 (全部实测) + +```bash +# ── 4.1 Docker 与内核调优: 沿用 livekit 项目成果 +# (/etc/sysctl.d/99-livekit-sip.conf 已应用, 见 §8; asterisk 运行时已下线) + +# ── 4.2 部署栈 ───────────────────────────────────────────────────── +mkdir -p /opt/freeswitch-esl && cd /opt/freeswitch-esl # = 仓库 deploy/ 目录 +./scripts/setup-conf.sh # 生成 etc-fs/ (镜像 vanilla 配置 + 4 处补丁, 见踩坑) +docker compose up -d # fs1 + mock-provider +docker compose --profile scale up -d # (可选) fs2 第二节点 + +# ── 4.3 验证 ─────────────────────────────────────────────────────── +./scripts/health.sh # 容器 + ESL status + gateway 状态 + 活动呼叫 +./scripts/call.sh +15105550123 12 1 # 端到端外呼 (节点1) +./scripts/call.sh +15105550123 10 2 # 端到端外呼 (节点2, 验证扩展) +``` + +配置仅 2 个小文件 (conf/): `event_socket.conf.xml` (ESL 0.0.0.0:8021 + 密码 + rfc1918.auto ACL)、`mock-trunk.xml` (sofia 网关: 无注册 + OPTIONS ping 25s 探活); 其余由 `setup-conf.sh` 从镜像 vanilla 配置生成并打补丁。 + +⚠ 踩坑1 (**最关键**): safarov 镜像 entrypoint 在 `/etc/freeswitch/freeswitch.xml` 不存在时从 vanilla 种子整目录 cp —— **按文件 RO 挂载会报 "Read-only file system" 且配置落空**; 必须整目录挂载 (`./etc-fs:/etc/freeswitch`), 由 `setup-conf.sh` 生成 (每次全量重建, 幂等)。 +⚠ 踩坑2: vanilla `event_socket` 的 `listen-ip=::` 在本容器 getaddrinfo 失败 → mod_event_socket 起不来; 改 `0.0.0.0`。 +⚠ 踩坑3: vanilla `vars.xml` 的 `stun-set` 在配置解析期做外网 STUN 查询, 不通则 `Invalid ext-sip-ip` → **mod_sofia 整体加载失败**; setup-conf.sh 已剥离 stun-set 与 ext-sip-ip/ext-rtp-ip。 +⚠ 踩坑4 (ESL ACL, 折腾最久): 不设 `apply-inbound-acl` → 默认 loopback.auto, 容器网内 agent 被拒 (`text/rude-rejection`); `localnet.auto` 不含 127.0.0.1 → 容器内 fs_cli 被拒; **自定义 acl.conf.xml 列表在此镜像实测不生效** (127.0.0.1 与 eth0 均被拒)。解: 内置 `rfc1918.auto` (10/8+172.16/12+192.168/16) — agent 与 fs_cli 均放行, fs_cli 需 `-H $(hostname -i)` 走 eth0 源地址。 +⚠ 踩坑5: 镜像缺 CA 证书, mod_signalwire 每分钟刷 Curl 77 错误日志 → 从 modules.conf.xml 删除加载行。 +⚠ 踩坑6: fs_cli 默认连 localhost+ClueCon, 需显式 `-H -P 8021 -p <密码>`; 且 count 类命令输出**首行是空行**, 解析要过滤。 +⚠ 踩坑7: FS 启动后第一个 originate 可能因网关冷启动未就绪失败; mock-agent 对腿2 已做 3 次重试。 + +## 5. 外呼链路验证 (实测输出) + +`./scripts/call.sh +15105550123 12 1`: +``` +[agent] agent up: sip=udp/5062 rtp=udp/40001 ip=172.18.0.4 +[agent] esl connected: freeswitch1:8021 +[agent] INVITE from FS 172.18.0.3:5080 call_id=7bdc5fbc-... ← FS originate 腿1 到 agent +[agent] leg answered (ACK), fs rtp=('172.18.0.3', 18364) codec=PCMU +[agent] leg1 (agent UAS) up: +OK agent-1788159625-782 +[agent] leg2 (callee) answered: +OK callee-1788159625-782 ← 经 mock-trunk 外呼被叫 +[agent] bridged [agent-... + callee-...], streaming 12.0s +[agent] RX frames=500 avg_rms=8362 +RESULT number=+15105550123 rx_frames=536 rx_avg_rms=8246 tx_frames=568 bidirectional=YES +``` +mock-provider (被叫侧) 日志: +``` +INCOMING_INVITE from=172.18.0.3:5080 uri="sip:+15105550123@mock-provider:5060" +ANSWERED media=172.18.0.2:40000 codec=PCMU peer=172.18.0.3:23888 ← SDP 协商 PCMU +ACK confirmed, RTP bridging +CALL_DONE reason=bye dur=13.5s sent_rtp=570 recv_rtp=533 recv_avg_amplitude=8236 +``` +**双向语音证明**: agent RX=534 帧 (被叫→ASR 方向, RMS 8270 = 440Hz 检测到) 与 provider recv_rtp=533 (LLM/TTS→被叫方向) **计数完全对称**; agent TX=568 ↔ provider sent_rtp=570。挂断后 `show calls count`=0, 无泄漏。换真实 LLM/ASR 只需替换 mock-agent 的 TX 音源与 RX 消费循环。 + +## 6. 运行状态检测 + +`./scripts/health.sh` (实测输出): +``` +== 容器状态 fs1 / fs2 (healthy) / fs-mock-provider Up +== ESL status PASS fs1 (UP, FreeSWITCH 1.10.12) PASS fs2 +== sofia 网关 PASS fs1 mock-trunk State=NOREG PASS fs2 (NOREG=无注册模式+ping 探活, 正常态) +== 活动呼叫 PASS fs1 active calls=0 +result: ok=5 fail=0 +``` +持续监控: `watch -n5 ./scripts/health.sh`; 呼叫级状态: provider 日志 `CALL_DONE` 行 (时长/双向计数); 容器级: docker healthcheck (compose 已配 fs_cli status)。层级与 livekit/asterisk 版对齐: 容器 + API 功能 + 业务功能三层。 + +## 7. 水平扩展实证 (双节点, 零配置变更) + +**① 加节点即扩容**: `docker compose --profile scale up -d` 拉起 fs2, 共用同一套 etc-fs, 无任何既有节点配置改动。 + +**② 3 路并发分摊双节点 (发起侧调度)**: +``` +$ (./scripts/call.sh +15105550123 10 1 &) (./scripts/call.sh +15105550123 10 1 &) (./scripts/call.sh +15105550123 10 2 &) +RESULT ... rx_frames=440 rx_avg_rms=8245 tx_frames=471 bidirectional=YES +RESULT ... rx_frames=438 rx_avg_rms=8193 tx_frames=471 bidirectional=YES +RESULT ... rx_frames=440 rx_avg_rms=8178 tx_frames=470 bidirectional=YES + +$ docker logs fs-mock-provider | grep INCOMING | grep -oE "from=[0-9.]+" | sort | uniq -c + 2 from=172.18.0.3 ← fs1 处理 2 路 (源 IP = fs1) + 1 from=172.18.0.4 ← fs2 处理 1 路 (源 IP = fs2) + +$ (呼中 t=8s) fs1: 2 calls / 4 channels fs2: 1 call / 2 channels ← 每桥接呼叫=2通道 +$ (provider) CALL_DONE sent_rtp=473 recv_rtp=439 amplitude=8174 ← 双向对称 +``` +→ 加副本即扩容、发起侧调度、双节点各自满血双向, 三项均实证。 +资源基线 (3 并发): fs1 0.3%CPU/48M, fs2 0.3%CPU/44M — 2vCPU 单节点距容量上限还有两个数量级余量。 + +## 8. 系统调优 + +**内核** (沿用已应用的 `/etc/sysctl.d/99-livekit-sip.conf`, RTP/UDP 通用, 无需改动): +```conf +net.core.rmem_max = 16777216 # UDP 收缓冲上限 (16M, 默认 208K 高并发必丢包) +net.core.wmem_max = 16777216 +net.core.rmem_default = 1048576 +net.core.wmem_default = 1048576 +net.core.netdev_max_backlog = 4096 +net.ipv4.ip_local_port_range = 10000 65000 # 出向 SIP/RTP 源端口 +net.ipv4.udp_mem = 8388608 12582912 16777216 # 全局 UDP 页缓存 (3.5G 内存安全值) +``` +验证: `nstat -az | grep -i udp` (`UdpRcvbufErrors` 应为 0)。 + +**组件级**: +- RTP 段: `switch.conf.xml` 的 `rtp-start-port/rtp-end-port` (vanilla 默认 16384-32768 已注释未改); 生产按并发收窄或分段, 端口不够优先加节点 (见 §7)。 +- ESL: 高并发控制面改 `bgapi` (异步作业) 或 esl 连接池; 单连接串行 `api` 是同步阻塞的 (本文 mock 场景够用)。`# ponytail: 单 ESL 连接串行 api, >100 并发呼叫控制需 bgapi/连接池`。 +- sofia 网关: `ping=25` OPTIONS 探活快速摘除故障 trunk; 网关冷启动有秒级窗口 (mock-agent 已重试兜底)。 +- 容器: `restart: unless-stopped` + healthcheck 已配; 双节点各 ~45M 内存, 3.5G 机器无需 mem limit; `cap_add: SYS_NICE` (FS 定时器优先级)。 +- 监控: health.sh 三层 + `show calls count` 告警 (骤降); FS 文件日志 `/var/log/freeswitch/freeswitch.log`; 告警项: 容器重启、gateway State 异常、UdpRcvbufErrors>0。 + +## 9. 生产化清单 (Mock → 真实运营商) + +1. trunk: mock-trunk.xml 加 `username/password` (digest 注册) 或 IP 白名单模式, proxy 换运营商 SIP 域名/IP; 我方 5060/udp(external profile 5080 改)、RTP 段公网放行 (阿里云安全组)。 +2. ESL 安全: 8021/8022 仅对 agent 网段放行; 换强密码; 保持 rfc1918.auto ACL (生产可再收紧到容器网段); **ESL 等同 root shell, 公网暴露=事故**。 +3. NAT: 阿里云 ECS 1:1 NAT — external profile 配 `ext-sip-ip/ext-rtp-ip` 为公网 IP (本文已剥离的参数届时显式回填, 不要用 stun); agent 侧 SIP/RTP 地址用宿主可达地址。 +4. AI 侧: mock-agent 替换为真实 STT/LLM/TTS (RX 帧喂识别、TX 换合成音频; SIP UAS + ESL 接口不变)。 +5. 高可用: 节点 ≥2 (已验证); 调度侧带健康检查摘除故障节点; 需要呼入/注册时前置 Kamailio dispatcher。 diff --git a/freeswitch/deploy/conf/event_socket.conf.xml b/freeswitch/deploy/conf/event_socket.conf.xml new file mode 100644 index 0000000..b5e77aa --- /dev/null +++ b/freeswitch/deploy/conf/event_socket.conf.xml @@ -0,0 +1,15 @@ + + + + + + + + + + diff --git a/freeswitch/deploy/conf/mock-trunk.xml b/freeswitch/deploy/conf/mock-trunk.xml new file mode 100644 index 0000000..58dc2a5 --- /dev/null +++ b/freeswitch/deploy/conf/mock-trunk.xml @@ -0,0 +1,10 @@ + + + + + + + + + + diff --git a/freeswitch/deploy/docker-compose.yml b/freeswitch/deploy/docker-compose.yml new file mode 100644 index 0000000..a58d0bb --- /dev/null +++ b/freeswitch/deploy/docker-compose.yml @@ -0,0 +1,55 @@ +name: freeswitch-esl + +services: + freeswitch1: + image: safarov/freeswitch:latest + container_name: fs1 + volumes: + - ./etc-fs:/etc/freeswitch + ports: ["8021:8021"] + cap_add: [SYS_NICE] + healthcheck: + test: ["CMD-SHELL", "fs_cli -H $(hostname -i) -P 8021 -p esl_mock_a95a037399153e9092b653fc -x status | grep -q '^UP'"] + interval: 15s + timeout: 5s + retries: 3 + start_period: 30s + restart: unless-stopped + networks: [fsnet] + + # 第二个 FreeSWITCH 节点: 无共享状态, 出站外呼各自独立 (水平扩展验证) + freeswitch2: + image: safarov/freeswitch:latest + container_name: fs2 + volumes: + - ./etc-fs:/etc/freeswitch + ports: ["8022:8021"] + cap_add: [SYS_NICE] + healthcheck: + test: ["CMD-SHELL", "fs_cli -H $(hostname -i) -P 8021 -p esl_mock_a95a037399153e9092b653fc -x status | grep -q '^UP'"] + interval: 15s + timeout: 5s + retries: 3 + start_period: 30s + restart: unless-stopped + networks: [fsnet] + profiles: [scale] + + # Mock SIP 运营商 (UAS): 应答 FreeSWITCH 的出站 INVITE + mock-provider: + image: python:3.12-alpine + container_name: fs-mock-provider + working_dir: /app + command: python mock/mock-provider.py + volumes: + - ./mock:/app/mock:ro + environment: + SIP_PORT: "5060" + RTP_PORT: "40000" + RING_DELAY: "1.5" + restart: unless-stopped + networks: [fsnet] + +networks: + fsnet: + driver: bridge diff --git a/freeswitch/deploy/mock/mock-agent.py b/freeswitch/deploy/mock/mock-agent.py new file mode 100644 index 0000000..591e992 --- /dev/null +++ b/freeswitch/deploy/mock/mock-agent.py @@ -0,0 +1,316 @@ +#!/usr/bin/env python3 +"""Mock AI Agent — FreeSWITCH/ESL 版. + +方案: FreeSWITCH 无 ARI externalMedia 式外挂媒体 API, 标准做法是本 agent 自身充当 +SIP UAS 端点 (等价于 asterisk 版的 externalMedia 腿), 控制面走 ESL: + 0. ESL TCP 连接 fs:8021 (auth) + 1. 本侧 SIP UAS 监听 (5062) + RTP 监听 (40001) + 2. ESL originate 腿1: sofia/external/agent@<本容器>:5062 &park() ← 本侧 UAS 应答 + 3. ESL originate 腿2: sofia/gateway/mock-trunk/<被叫> &park() ← mock-provider 应答 + 4. ESL uuid_bridge 腿1 腿2 → 桥接, RTP 经 FreeSWITCH 双向 + 5. RTP: 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. ESL uuid_kill 双腿 → BYE, 输出双向统计 +纯 stdlib (asyncio + socket)。换真实 LLM/ASR: 替换 TX 音源与 RX 消费即可。 +用法: mock-agent.py --esl-host fs1 --sip-host <本容器名>:5062 \ + --number +15105550123 [--duration 15] +""" +import argparse, asyncio, math, os, random, socket, struct, time + +def log(*a): print(f"[agent {time.strftime('%H:%M:%S')}]", *a, flush=True) + +# ---------- G.711 编码 ---------- +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 alaw_encode(s: int) -> int: + s = max(-32635, min(32635, s)); sign = 0x80 if s < 0 else 0 + if sign: s = -s + e = 7 + for i in range(7, -1, -1): + if s & (0x40 << i): e = i; break + m = (s >> (e + 3)) & 0x0F + return ((sign | (e << 4) | m) ^ 0x55) & 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 alaw_decode(a: int) -> int: + a ^= 0x55 + t = (a & 0x0F) << 4 + seg = (a & 0x70) >> 4 + if seg == 0: t += 8 + else: + t += 0x108 + if seg > 1: t <<= (seg - 1) + return -t if a & 0x80 else t + +def tone_payload(codec: str, ts: float) -> bytes: + """20ms / 160 samples @8kHz, 440Hz 500ms-on 500ms-off (模拟 TTS)""" + enc = ulaw_encode if codec == "PCMU" else alaw_encode + 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(enc(v)) + return bytes(out) + +# ---------- SIP 报文 (与 mock-provider 同源) ---------- +def parse_msg(data: bytes): + head, _, body = data.decode("utf-8", "replace").partition("\r\n\r\n") + lines = head.split("\r\n") + headers = [] + for ln in lines[1:]: + n, _, v = ln.partition(":") + headers.append((n.strip().lower(), v.strip())) + return lines[0], headers, body + +def hget(headers, name): + for n, v in headers: + if n == name: return v + return None + +def build_resp(first, headers, code, reason, tag=None, extra=None, body=""): + out = [f"SIP/2.0 {code} {reason}"] + for n, v in headers: + if n == "via": out.append(f"Via: {v}") + out.append(f"From: {hget(headers,'from')}") + to = hget(headers, "to") + if tag and "tag=" not in to: to = f"{to};tag={tag}" + out.append(f"To: {to}") + out.append(f"Call-ID: {hget(headers,'call-id')}") + out.append(f"CSeq: {hget(headers,'cseq')}") + if extra: + out += extra + if body: + out.append("Content-Type: application/sdp") + out.append(f"Content-Length: {len(body)}") + return ("\r\n".join(out) + "\r\n\r\n" + body).encode() + +def parse_sdp_offer(body: str): + ip, port, pts = None, None, [] + for ln in body.splitlines(): + if ln.startswith("c=IN IP4 "): ip = ln.split()[2] + elif ln.startswith("m=audio "): + parts = ln.split(); port = int(parts[1]); pts = [p for p in parts[3:] if p.isdigit()] + return ip, port, pts + +def pick_codec(pts): + if "0" in pts: return "0", "PCMU" + if "8" in pts: return "8", "PCMA" + return (pts[0], "PCMU") if pts else ("0", "PCMU") + +# ---------- SIP UAS (被 FreeSWITCH originate 的"externalMedia 腿") ---------- +class SipUas(asyncio.DatagramProtocol): + """应答 FS 的 INVITE (200+SDP), 处理 ACK/BYE/OPTIONS/re-INVITE/重传.""" + def __init__(self, my_ip, sip_port, rtp_port): + self.my_ip, self.sip_port, self.rtp_port = my_ip, sip_port, rtp_port + self.transport = None + self.tag = os.urandom(5).hex() + self.cid = None + self.fs_rtp = None # FS 侧 RTP 地址 (INVITE SDP) + self.pt, self.codec = "0", "PCMU" + self.confirmed = False + self.ended = asyncio.Event() + + def connection_made(self, transport): + self.transport = transport + + def datagram_received(self, data, addr): + first, headers, body = parse_msg(data) + method = first.split()[0] if first else "?" + if method == "INVITE": + cid = hget(headers, "call-id") + new_call = self.cid is None + if new_call: + self.cid = cid + log(f"INVITE from FS {addr[0]}:{addr[1]} call_id={cid}") + ip, port, pts = parse_sdp_offer(body) + if ip and port: + self.fs_rtp = (ip, port) # re-INVITE 时同步更新 + if new_call: + self.pt, self.codec = pick_codec(pts) + # 重传与 re-INVITE 统一: 按当前请求头重建 200 OK (CSeq 正确) + sdp = (f"v=0\r\no=agent {random.randint(1,10**9)} {random.randint(1,10**9)} IN IP4 {self.my_ip}\r\n" + f"s=agent\r\nc=IN IP4 {self.my_ip}\r\nt=0 0\r\n" + f"m=audio {self.rtp_port} RTP/AVP {self.pt}\r\n" + f"a=rtpmap:{self.pt} {self.codec}/8000\r\na=sendrecv\r\n") + self.transport.sendto(build_resp( + first, headers, 200, "OK", tag=self.tag, + extra=[f"Contact: ", + "User-Agent: MockAgent/1.0"], body=sdp), addr) + elif method == "ACK": + if not self.confirmed: + self.confirmed = True + log(f"leg answered (ACK), fs rtp={self.fs_rtp} codec={self.codec}") + elif method == "BYE": + self.transport.sendto(build_resp(first, headers, 200, "OK", tag=self.tag), addr) + self.ended.set() + elif method == "OPTIONS": + self.transport.sendto(build_resp(first, headers, 200, "OK", tag=self.tag), addr) + +# ---------- RTP 媒体 ---------- +class Rx(asyncio.DatagramProtocol): + def __init__(self, stats): + self.stats = stats + + def datagram_received(self, data, addr): + if len(data) < 12: return + pt = data[1] & 0x7F + self.stats["rx"] += 1 + payload = data[12:] + dec = [(ulaw_decode(b) if pt == 0 else alaw_decode(b)) for b in payload[:40]] + self.stats["amp_sum"] += sum(abs(x) for x in dec) / max(1, len(dec)) + self.stats["tx_to"] = addr # 始终跟随最新来源 (自适应媒体路径) + if self.stats["rx"] % 250 == 0: + log(f"RX frames={self.stats['rx']} avg_rms={self.stats['amp_sum']/self.stats['rx']:.0f}") + +# ---------- ESL 客户端 (极简: auth + api, 同步命令式) ---------- +class Esl: + def __init__(self, host, port, password): + self.host, self.port, self.password = host, port, password + + async def connect(self): + self.r, self.w = await asyncio.open_connection(self.host, self.port) + m = await self._read() + if m.get("content-type") != "auth/request": + raise ConnectionError(f"unexpected esl greeting: {m}") + self.w.write(f"auth {self.password}\n\n".encode()) + await self.w.drain() + m = await self._read() + if "+OK" not in m.get("reply-text", ""): + raise ConnectionError(f"esl auth failed: {m}") + + async def _read(self): + headers = {} + while True: + line = await self.r.readline() + if not line: raise ConnectionError("esl closed") + line = line.decode("utf-8", "replace").rstrip("\r\n") + if not line: break + k, _, v = line.partition(":") + headers[k.strip().lower()] = v.strip() + n = int(headers.get("content-length", 0) or 0) + headers["_body"] = (await self.r.readexactly(n)).decode("utf-8", "replace") if n else "" + return headers + + async def api(self, cmd): + self.w.write(f"api {cmd}\n\n".encode()) + await self.w.drain() + return (await self._read())["_body"].strip() + + async def close(self): + try: self.w.close() + except Exception: pass + +# ---------- 主流程 ---------- +async def run(args): + my_ip = socket.gethostbyname(socket.gethostname()) + sip_port = int(args.sip_host.rsplit(":", 1)[1]) + suffix = f"{int(time.time())}-{random.randint(100, 999)}" + agent_uuid, callee_uuid = f"agent-{suffix}", f"callee-{suffix}" + tmo = int(args.ring_timeout) + + stats = {"rx": 0, "amp_sum": 0.0, "tx": 0, "tx_to": None, "codec": "PCMU"} + loop = asyncio.get_running_loop() + uas_t, uas = await loop.create_datagram_endpoint( + lambda: SipUas(my_ip, sip_port, args.rtp_port), local_addr=("0.0.0.0", sip_port)) + rtp_t, _ = await loop.create_datagram_endpoint( + lambda: Rx(stats), local_addr=("0.0.0.0", args.rtp_port)) + log(f"agent up: sip=udp/{sip_port} rtp=udp/{args.rtp_port} ip={my_ip}") + + esl = Esl(args.esl_host, args.esl_port, args.esl_password) + await esl.connect() + log(f"esl connected: {args.esl_host}:{args.esl_port}") + + def fail(msg, detail=None): + log(f"FAIL: {msg}", detail or "") + raise SystemExit(1) + + try: + # 腿1: 本 agent (SIP UAS, 等价 externalMedia): 应答后 park + r = await esl.api( + f"originate {{origination_uuid={agent_uuid}," + f"origination_caller_id_number={args.caller_id},originate_timeout={tmo}}}" + f"sofia/external/agent@{args.sip_host} &park()") + if not r.startswith("+OK"): fail("originate agent leg", r) + log(f"leg1 (agent UAS) up: {r}") + + # 腿2: 被叫经 mock-trunk 网关 (网关冷启动自动重试) + for attempt in range(3): + r = await esl.api( + f"originate {{origination_uuid={callee_uuid}," + f"origination_caller_id_number={args.caller_id},originate_timeout={tmo}}}" + f"sofia/gateway/mock-trunk/{args.number} &park()") + if r.startswith("+OK"): break + log(f"leg2 originate retry {attempt + 1}: {r}") + await asyncio.sleep(2) + if not r.startswith("+OK"): fail("originate callee leg", r) + log(f"leg2 (callee) answered: {r}") + + stats["codec"] = uas.codec + if uas.fs_rtp and not stats["tx_to"]: + stats["tx_to"] = uas.fs_rtp + log(f"fs rtp peer (INVITE SDP): {uas.fs_rtp[0]}:{uas.fs_rtp[1]}") + + # 桥接双腿 → RTP 经 FS 双向 + r = await esl.api(f"uuid_bridge {agent_uuid} {callee_uuid}") + if not r.startswith("+OK"): fail("uuid_bridge", r) + log(f"bridged [{agent_uuid} + {callee_uuid}], streaming {args.duration}s " + f"(RX=ASR mock, TX=440Hz LLM/TTS mock)") + + # 媒体收发 + async def tx(): + seq = random.randint(0, 65535); ts = random.randint(0, 10**6) + ssrc = random.randint(1, 2**31) + pt = 0 if stats["codec"] == "PCMU" else 8 + t0 = time.time() + while time.time() - t0 < args.duration: + if stats["tx_to"]: + hdr = struct.pack("!BBHII", 0x80, pt, seq & 0xFFFF, + ts & 0xFFFFFFFF, ssrc) + rtp_t.sendto(hdr + tone_payload(stats["codec"], time.time() - t0), + stats["tx_to"]) + stats["tx"] += 1 + seq += 1; ts += 160 + await asyncio.sleep(0.02) + await asyncio.wait_for(tx(), timeout=args.duration + 5) + finally: + for u in (callee_uuid, agent_uuid): + try: await esl.api(f"uuid_kill {u}") + except Exception: pass + await esl.close() + uas_t.close(); rtp_t.close() + await asyncio.sleep(0.3) # 等 BYE 应答落地 + + 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) + +def main(): + ap = argparse.ArgumentParser() + ap.add_argument("--esl-host", default="fs1") + ap.add_argument("--esl-port", type=int, default=8021) + ap.add_argument("--esl-password", default="esl_mock_a95a037399153e9092b653fc") + ap.add_argument("--sip-host", required=True, help="本侧 SIP 地址 host:port (供 FS INVITE)") + ap.add_argument("--rtp-port", type=int, default=40001) + ap.add_argument("--number", default="+15105550123") + ap.add_argument("--caller-id", default="+15109990001") + ap.add_argument("--duration", type=float, default=15) + ap.add_argument("--ring-timeout", type=float, default=30) + args = ap.parse_args() + asyncio.run(run(args)) + +if __name__ == "__main__": + main() diff --git a/freeswitch/deploy/mock/mock-provider.py b/freeswitch/deploy/mock/mock-provider.py new file mode 100755 index 0000000..3afc5c7 --- /dev/null +++ b/freeswitch/deploy/mock/mock-provider.py @@ -0,0 +1,244 @@ +#!/usr/bin/env python3 +"""Mock SIP 运营商 (UAS): 接收 Asterisk(PJSIP) 出站 INVITE, 振铃->应答, 双向收发 PCMU/PCMA RTP. +仅 stdlib, 无依赖。一个 RTP 端口复用所有呼叫(按来源 IP 匹配呼叫)。 +用法: python3 mock-provider.py (环境变量 SIP_PORT/RTP_PORT/RING_DELAY) +""" +import asyncio, math, os, random, socket, struct, time + +SIP_PORT = int(os.environ.get("SIP_PORT", "5060")) +RTP_PORT = int(os.environ.get("RTP_PORT", "40000")) +RING_DELAY = float(os.environ.get("RING_DELAY", "1.5")) +MY_IP = socket.gethostbyname(socket.gethostname()) + +def log(*a): print(f"[provider {time.strftime('%H:%M:%S')}]", *a, flush=True) + +# ---------- G.711 编码 ---------- +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 alaw_encode(s: int) -> int: + s = max(-32635, min(32635, s)); sign = 0x80 if s < 0 else 0 + if sign: s = -s + e = 7 + for i in range(7, -1, -1): + if s & (0x40 << i): e = i; break + m = (s >> (e + 3)) & 0x0F + return ((sign | (e << 4) | m) ^ 0x55) & 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(codec: str, ts: float) -> bytes: + """20ms / 160 samples @8kHz, 440Hz 500ms-on 500ms-off""" + 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) if codec == "PCMU" else alaw_encode(v)) + return bytes(out) + +# ---------- SIP 报文 ---------- +def parse_msg(data: bytes): + head, _, body = data.decode("utf-8", "replace").partition("\r\n\r\n") + lines = head.split("\r\n") + headers = [] + for ln in lines[1:]: + n, _, v = ln.partition(":") + headers.append((n.strip().lower(), v.strip())) + return lines[0], headers, body + +def hget(headers, name): + for n, v in headers: + if n == name: return v + return None + +def build_resp(first, headers, code, reason, tag=None, extra=None, body=""): + out = [f"SIP/2.0 {code} {reason}"] + for n, v in headers: + if n == "via": out.append(f"Via: {v}") + out.append(f"From: {hget(headers,'from')}") + to = hget(headers, "to") + if tag and "tag=" not in to: to = f"{to};tag={tag}" + out.append(f"To: {to}") + out.append(f"Call-ID: {hget(headers,'call-id')}") + out.append(f"CSeq: {hget(headers,'cseq')}") + if extra: + out += extra + if body: + out.append(f"Content-Type: application/sdp") + out.append(f"Content-Length: {len(body)}") + return ("\r\n".join(out) + "\r\n\r\n" + body).encode() + +def parse_sdp_offer(body: str): + """返回 (对端媒体IP, 对端媒体端口, payload类型列表)""" + ip, port, pts = None, None, [] + for ln in body.splitlines(): + if ln.startswith("c=IN IP4 "): ip = ln.split()[2] + elif ln.startswith("m=audio "): + parts = ln.split(); port = int(parts[1]); pts = [p for p in parts[3:] if p.isdigit()] + return ip, port, pts + +def sdp_answer(codec_pt: str, codec_name: str) -> str: + sid = random.randint(1, 10**9) + return (f"v=0\r\no=mock {sid} {sid+1} IN IP4 {MY_IP}\r\ns=mock\r\n" + f"c=IN IP4 {MY_IP}\r\nt=0 0\r\n" + f"m=audio {RTP_PORT} RTP/AVP {codec_pt}\r\n" + f"a=rtpmap:{codec_pt} {codec_name}/8000\r\na=sendrecv\r\n") + +# ---------- 呼叫对象 ---------- +class Call: + def __init__(self, first, headers, addr): + self.id = hget(headers, "call-id") + self.tag = os.urandom(5).hex() + self.first, self.headers, self.addr = first, headers, addr + self.peer_ip = self.peer_port = None + self.codec = "PCMU"; self.pt = "0" + self.ok_raw = None; self.confirmed = False; self.closed = False + self.sent = self.recv = self.recv_bytes = self.recv_amp_sum = 0 + self.t0 = time.time() + self.last_resp = None + +calls: dict[str, Call] = {} +rtp_sock = None + +def pick_codec(pts): + if "0" in pts: return "0", "PCMU" + if "8" in pts: return "8", "PCMA" + return (pts[0], "PCMU") if pts else ("0", "PCMU") + +async def ringing_then_answer(call, sip_t): + await asyncio.sleep(RING_DELAY) + if call.closed: return + body = sdp_answer(call.pt, call.codec) + call.ok_raw = build_resp(call.first, call.headers, 200, "OK", tag=call.tag, + extra=[f"Contact: ", + "User-Agent: MockProvider/1.0"], body=body) + call.last_resp = call.ok_raw + sip_t.sendto(call.ok_raw, call.addr) + log(f"ANSWERED call_id={call.id} media={MY_IP}:{RTP_PORT} codec={call.codec} peer={call.peer_ip}:{call.peer_port}") + start_rtp(call) + for _ in range(5): # 重传 200 OK 直到 ACK (UDP 可靠性) + await asyncio.sleep(1.0) + if call.confirmed or call.closed: break + sip_t.sendto(call.ok_raw, call.addr) + +def start_rtp(call): + async def sender(): + seq = random.randint(0, 65535); ts = random.randint(0, 10**6) + ssrc = random.randint(1, 2**31) + pt = 0 if call.codec == "PCMU" else 8 + t0 = time.time() + while not call.closed: + now = time.time() - t0 + payload = tone_payload(call.codec, now) + hdr = struct.pack("!BBHII", 0x80, pt, seq & 0xFFFF, ts & 0xFFFFFFFF, ssrc) + try: + rtp_sock.sendto(hdr + payload, (call.peer_ip, call.peer_port)) + except OSError: + pass + call.sent += 1; seq += 1; ts += 160 + await asyncio.sleep(0.02) + asyncio.create_task(sender()) + +def close_call(call, reason): + if call.closed: return + call.closed = True + dur = time.time() - call.t0 + avg = call.recv_amp_sum / call.recv if call.recv else 0 + log(f"CALL_DONE call_id={call.id} reason={reason} dur={dur:.1f}s " + f"sent_rtp={call.sent} recv_rtp={call.recv} recv_bytes={call.recv_bytes} recv_avg_amplitude={avg:.0f}") + calls.pop(call.id, None) + +def on_sip(data, addr, sip_t): + first, headers, body = parse_msg(data) + method = first.split()[0] if first else "?" + cid = hget(headers, "call-id") + + if method == "INVITE": + call = calls.get(cid) + if call: # 重传: 回最后一条响应 + if call.last_resp: sip_t.sendto(call.last_resp, addr) + return + call = Call(first, headers, addr) + call.peer_ip, call.peer_port, pts = parse_sdp_offer(body) + call.pt, call.codec = pick_codec(pts) + calls[cid] = call + log(f"INCOMING_INVITE from={addr[0]}:{addr[1]} uri=\"{first.split()[1]}\" call_id={cid}") + sip_t.sendto(build_resp(first, headers, 100, "Trying"), addr) + sip_t.sendto(build_resp(first, headers, 180, "Ringing"), addr) + call.last_resp = build_resp(first, headers, 180, "Ringing") + asyncio.create_task(ringing_then_answer(call, sip_t)) + + elif method == "ACK": + call = calls.get(cid) + if call and not call.confirmed: + call.confirmed = True + log(f"ACK call_id={cid} confirmed, RTP bridging") + + elif method == "BYE": + call = calls.get(cid) + sip_t.sendto(build_resp(first, headers, 200, "OK", tag=call.tag if call else None), addr) + if call: close_call(call, "bye") + + elif method == "CANCEL": + call = calls.get(cid) + sip_t.sendto(build_resp(first, headers, 487, "Request Terminated", + tag=call.tag if call else None), addr) + if call: close_call(call, "cancelled") + + elif method == "OPTIONS": + sip_t.sendto(build_resp(first, headers, 200, "OK", tag="mock"), addr) + +class RtpProto(asyncio.DatagramProtocol): + def datagram_received(self, data, addr): + if len(data) < 12: return + pt = data[1] & 0x7F + payload = data[12:] + # 优先 ip:port 精确匹配 (同 IP 多并发呼叫时按 RTP 端口区分), 兜底仅 IP + for call in list(calls.values()): + if not call.closed and call.peer_ip == addr[0] and call.peer_port == addr[1]: + self._tally(call, pt, payload); return + for call in list(calls.values()): + if not call.closed and call.peer_ip == addr[0]: + self._tally(call, pt, payload); return + + @staticmethod + def _tally(call, pt, payload): + call.recv += 1; call.recv_bytes += len(payload) + 12 + if payload and pt in (0, 8): + dec = [ulaw_decode(b) for b in payload[:40]] + call.recv_amp_sum += sum(abs(x) for x in dec) / len(dec) + +class SipProto(asyncio.DatagramProtocol): + def __init__(self): self.t = None + def connection_made(self, transport): self.t = transport + def datagram_received(self, data, addr): + if self.t: on_sip(data, addr, self.t) + +async def main(): + global rtp_sock + loop = asyncio.get_running_loop() + sip_t, _ = await loop.create_datagram_endpoint( + lambda: SipProto(), local_addr=("0.0.0.0", SIP_PORT)) + rtp_t, _ = await loop.create_datagram_endpoint( + RtpProto, local_addr=("0.0.0.0", RTP_PORT)) + rtp_sock = rtp_t + log(f"mock provider up: sip=udp/{SIP_PORT} rtp=udp/{RTP_PORT} ip={MY_IP}") + log("waiting for outbound INVITE from asterisk ...") + await asyncio.Event().wait() + +if __name__ == "__main__": + try: + asyncio.run(main()) + except KeyboardInterrupt: + pass diff --git a/freeswitch/deploy/scripts/call.sh b/freeswitch/deploy/scripts/call.sh new file mode 100755 index 0000000..3456818 --- /dev/null +++ b/freeswitch/deploy/scripts/call.sh @@ -0,0 +1,18 @@ +#!/bin/bash +# 端到端外呼验证: mock-agent(ESL+SIP UAS+RTP) → FreeSWITCH → mock-trunk → mock-provider(应答) → 双向 RTP 统计 +# 用法: ./call.sh [被叫号码] [时长秒] [freeswitch节点 1|2] +set -euo pipefail +cd "$(dirname "$0")/.." + +NUMBER="${1:-+15105550123}" +DURATION="${2:-15}" +NODE="${3:-1}" +NET="freeswitch-esl_fsnet" +NAME="mock-agent-$$" + +docker run --rm --name "$NAME" --network "$NET" \ + -v "$PWD/mock:/app/mock:ro" -w /app \ + python:3.12-alpine python mock/mock-agent.py \ + --esl-host "freeswitch${NODE}" --esl-port 8021 \ + --sip-host "${NAME}:5062" \ + --number "$NUMBER" --duration "$DURATION" diff --git a/freeswitch/deploy/scripts/health.sh b/freeswitch/deploy/scripts/health.sh new file mode 100755 index 0000000..a15de6c --- /dev/null +++ b/freeswitch/deploy/scripts/health.sh @@ -0,0 +1,53 @@ +#!/bin/bash +# 运行状态检测: 容器 + ESL 命令 + sofia 网关状态 + 活动呼叫数 +# 注: ESL ACL=rfc1918.auto 不含 loopback → fs_cli 必须 -H 走非 loopback 源地址; +# 且需显式 -P 8021 -p <密码> (fs_cli 默认 localhost+ClueCon) +set -uo pipefail +PASS="esl_mock_a95a037399153e9092b653fc" +ok=0; fail=0 +pass() { echo "PASS $1"; ok=$((ok+1)); } +faill() { echo "FAIL $1"; fail=$((fail+1)); } +running() { [ -n "$(docker ps -q -f name=$1)" ]; } +fsc() { # fsc <容器> + local ip + ip=$(docker exec "$1" hostname -i 2>/dev/null | awk '{print $1}') + docker exec "$1" fs_cli -H "$ip" -P 8021 -p "$PASS" "${@:2}" 2>/dev/null +} + +echo "== 容器状态" +docker ps --filter "name=fs" --format "{{.Names}}\t{{.Status}}" + +echo "== ESL (fs_cli status)" +for c in fs1 fs2; do + if running "$c"; then + out=$(fsc "$c" -x "status" | head -1) + if echo "$out" | grep -q "^UP"; then + pass "$c ESL status: ${out:0:60}" + else + faill "$c ESL status: ${out:-none}" + fi + else + echo "SKIP $c (未运行)" + fi +done + +echo "== sofia 网关 (mock-trunk)" +for c in fs1 fs2; do + if running "$c"; then + state=$(fsc "$c" -x "sofia status gateway mock-trunk" | awk -F'[:\t ]+' '/^State/{print $2}') + case "$state" in + REGED|NOREG) pass "$c gateway mock-trunk State=$state" ;; + *) faill "$c gateway mock-trunk State=${state:-none}" ;; + esac + fi +done + +echo "== 活动呼叫" +if running fs1; then + n=$(fsc fs1 -x "show calls count" | grep total | awk '{print $1}') + [ -n "$n" ] && pass "fs1 active calls=$n" || faill "fs1 active calls" +fi + +echo +echo "result: ok=$ok fail=$fail" +[ $fail -eq 0 ] diff --git a/freeswitch/deploy/scripts/setup-conf.sh b/freeswitch/deploy/scripts/setup-conf.sh new file mode 100755 index 0000000..c95ad11 --- /dev/null +++ b/freeswitch/deploy/scripts/setup-conf.sh @@ -0,0 +1,29 @@ +#!/bin/bash +# 生成 etc-fs 运行配置: 镜像 vanilla 配置 + 本仓库覆盖/补丁 +# 每次全量重建 (rm -rf → 拷 vanilla → 打补丁), 天然幂等, 不会叠加损坏 XML +# 产出: ../etc-fs/ (整目录挂载到容器 /etc/freeswitch, 见 docker-compose.yml) +set -euo pipefail +cd "$(dirname "$0")/.." +IMG="safarov/freeswitch:latest" + +# 1. 从镜像导出干净 vanilla 配置 +rm -rf etc-fs && mkdir -p etc-fs +docker run --rm --entrypoint sh -v "$PWD/etc-fs:/out" "$IMG" \ + -c "cp -a /usr/share/freeswitch/conf/vanilla/. /out/" + +# 2. ESL: 0.0.0.0:8021 + 自定义密码 + 内置 rfc1918.auto ACL +# (vanilla listen-ip=:: 在本容器 getaddrinfo 失败; ACL 详见文件内注释) +cp conf/event_socket.conf.xml etc-fs/autoload_configs/event_socket.conf.xml + +# 3. Mock 运营商 trunk 网关 (无注册 + OPTIONS ping 探活) +cp conf/mock-trunk.xml etc-fs/sip_profiles/external/mock-trunk.xml + +# 4. 去 stun: 依赖 (容器网内无 NAT; stun-set 在配置解析期做外网 STUN 查询, +# 不通则 Invalid ext-sip-ip → mod_sofia 加载失败) +sed -i '/cmd="stun-set"/d' etc-fs/vars.xml +sed -i '//d' etc-fs/autoload_configs/modules.conf.xml + +echo "etc-fs rebuilt (freeswitch.xml: $(ls etc-fs/freeswitch.xml))" diff --git a/freeswitch/deployment.md b/freeswitch/deployment.md new file mode 100644 index 0000000..edac4e6 --- /dev/null +++ b/freeswitch/deployment.md @@ -0,0 +1,25 @@ +目标机器：39.106.106.246 +登录用户: root + +任务目标：在 asterisk 运行时下线后，调研部署 FreeSWITCH 并以 ESL + Mock SIP 实现 SIP 对接与运行状态检测。 + +项目地址： https://github.com/signalwire/freeswitch (容器镜像 safarov/freeswitch = FreeSWITCH 1.10.12) +项目背景：使用 FreeSWITCH 对接 MOCK SIP 服务商，作为AI外呼机器人基础能力底座，使用MOCK对接LLM ASR实现双向语音通信, 本项目仅实现外呼功能，不实现多人房间与呼入功能调研。给出可实现最快水平扩展的结论方案并赋实践证据。给出详细的部署过程文档与系统调优方案。 + +--- + +## 交付状态 (2026-08-31 全部完成) + +| 目标 | 状态 | 证据 | +|---|---|---| +| asterisk 运行时下线 | ✅ | ast-* 容器清零, 8088/8089 端口释放; `/opt/asterisk-ari/` 交付物保留 | +| FreeSWITCH 部署 (双节点) | ✅ | fs1/fs2 healthy, ESL status UP ×2 (FreeSWITCH 1.10.12), health.sh ok=5 fail=0 | +| Mock SIP 对接 (外呼) | ✅ | ESL originate ×2 → INVITE→180→200 OK(PCMU)→ACK, gateway NOREG+ping 探活 ×2 | +| 双向语音 (Mock LLM/ASR) | ✅ | `RESULT ... bidirectional=YES` (agent RX 536帧 RMS 8246; provider recv_rtp=533, 双向计数对称) | +| 运行状态检测 | ✅ | `scripts/health.sh` 容器+ESL status+gateway+呼叫数, ok=5 fail=0 | +| 水平扩展结论+实证 | ✅ | 3 路并发分摊双节点 (INVITE 源 172.18.0.3×2 / 172.18.0.4×1, 呼中 fs1=2 calls/4ch, fs2=1/2), 零配置变更 | +| 部署文档+调优方案 | ✅ | 本仓库 `README.md` (§4 部署含 7 处踩坑, §8 sysctl 沿用已应用) | + +**调研要点 (vs Asterisk/ARI)**: FreeSWITCH 无 externalMedia 式外挂媒体 API —— mock-agent 自身充当 SIP UAS 端点 (FS originate 至 agent 并应答), 控制面走 ESL TCP (单连接命令式) 而非 ARI REST+WS; 水平扩展模型与 Asterisk 相同 (零共享状态节点复制, 不需要 livekit 那样的 Redis)。 + +**交付物**: 服务器 `/opt/freeswitch-esl/` (compose+conf+mock+scripts+etc-fs 生成器+README), 本地 `deploy/` 同源; 原始任务与完成状态见本文件, 详细文档见 `README.md`。