autonomic-bot
f283a371bb
Sanitized single-commit public mirror of recipe-maintainer. - Removed test-ssh/.testenv (live creds); added test-ssh/.testenv.example placeholders. - Removed plans/ and planned-updates/ (deployment-planning docs) so no client/ deployment domains appear in the public repo. - All other secret stores were already gitignored. - docs.coopcloud.tech retained as a submodule (public upstream).
414 lines
14 KiB
Python
414 lines
14 KiB
Python
#!/usr/bin/env python3
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"""
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WebRTC media flow test for La Suite Meet + LiveKit.
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Tests the full media path between two participants:
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1. Preliminary connectivity checks (TCP 7881, UDP 7882, UDP 443 TURN, WSS signaling)
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2. Both users authenticate via Keycloak OIDC
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3. User 1 creates a room via the Meet API
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4. User 2 joins the room
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5. Both connect to LiveKit via the SDK
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6. User 1 publishes a dummy audio track
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7. User 2 verifies audio frames arrive
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8. Clean up: disconnect both, delete the room
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This verifies:
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- WSS signaling through Traefik (port 7880)
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- ICE negotiation (direct or TURN relay)
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- Media transport over host-exposed ports (TCP 7881 / UDP 7882)
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- TURN relay via UDP 443 for clients behind restrictive NATs
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Network requirements:
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The test can succeed via direct ICE (ports 7881/7882) or via TURN relay
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(UDP 443). Users behind CGNAT/symmetric NAT will use TURN automatically.
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The preliminary checks report which paths are available.
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Prerequisites:
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pip install livekit requests
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Usage:
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python3 webrtc-media.py
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"""
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import asyncio
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import json
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import os
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import socket
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import ssl
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import struct
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import sys
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import time
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from pathlib import Path
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from urllib.parse import urlparse
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sys.path.insert(0, os.path.join(os.path.dirname(__file__), '..', '..', '..'))
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import requests
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from livekit import rtc
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from utils.tests.helpers import resolve_domain
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SAMPLE_RATE = 48000
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NUM_CHANNELS = 1
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SAMPLES_PER_CHANNEL = 480 # 10ms at 48kHz
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TIMEOUT = 45 # seconds to wait for audio frames (allow time for ICE TCP fallback)
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def load_credentials():
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"""Load Keycloak test credentials from domain-specific TOML file."""
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import tomllib
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from utils.tests.helpers import resolve_domain_suffix
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suffix = resolve_domain_suffix()
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creds_file = Path(__file__).resolve().parent.parent / f"keycloak-test-credentials.{suffix}.toml"
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if not creds_file.exists():
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print(f"FAIL: Credentials file not found: {creds_file}")
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print("Run setup_keycloak_integration.py first.")
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sys.exit(1)
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with open(creds_file, 'rb') as f:
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return tomllib.load(f)
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# ---------------------------------------------------------------------------
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# Preliminary connectivity checks
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# ---------------------------------------------------------------------------
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def check_tcp(host, port, timeout=5):
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"""Check if a TCP port is reachable."""
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s = socket.socket(socket.AF_INET, socket.SOCK_STREAM)
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s.settimeout(timeout)
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try:
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s.connect((host, port))
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s.close()
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return True
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except (socket.timeout, ConnectionRefusedError, OSError):
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return False
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def check_udp(host, port, timeout=5):
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"""Check if a UDP port accepts packets (send-only, best effort)."""
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s = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
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s.settimeout(timeout)
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try:
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s.sendto(b"\x00", (host, port))
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s.close()
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return True
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except OSError:
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return False
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def check_stun(host, port, timeout=5):
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"""Send a STUN Binding Request and verify we get a valid response.
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This proves the TURN/STUN server is actually listening and responding,
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not just that a UDP send succeeded (which almost always does).
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"""
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txn_id = os.urandom(12)
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# STUN Binding Request: type=0x0001, length=0, magic=0x2112A442
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msg = struct.pack("!HHI", 0x0001, 0, 0x2112A442) + txn_id
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s = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
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s.settimeout(timeout)
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try:
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s.sendto(msg, (host, port))
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data, _ = s.recvfrom(1024)
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except (socket.timeout, OSError):
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return False
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finally:
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s.close()
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if len(data) < 20:
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return False
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resp_type, _, magic = struct.unpack("!HHI", data[:8])
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resp_txn = data[8:20]
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# Valid STUN Binding Response: type=0x0101, correct magic + transaction ID
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return resp_type == 0x0101 and magic == 0x2112A442 and resp_txn == txn_id
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def check_wss(host, port=443, timeout=5):
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"""Check if a WSS (TLS) connection can be established."""
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s = socket.socket(socket.AF_INET, socket.SOCK_STREAM)
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s.settimeout(timeout)
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ctx = ssl.create_default_context()
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ss = ctx.wrap_socket(s, server_hostname=host)
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try:
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ss.connect((host, port))
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ss.close()
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return True
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except (socket.timeout, ConnectionRefusedError, OSError, ssl.SSLError):
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return False
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def run_connectivity_checks(meet_domain, livekit_domain):
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"""Run preliminary connectivity checks. Returns (all_critical_pass, turn_available)."""
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server_ip = socket.gethostbyname(meet_domain)
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livekit_ip = socket.gethostbyname(livekit_domain)
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print(f" Server IP: {server_ip} ({meet_domain})")
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print(f" LiveKit IP: {livekit_ip} ({livekit_domain})")
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all_pass = True
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turn_available = False
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# HTTPS / Traefik
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if check_tcp(server_ip, 443):
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print(f" PASS: TCP {server_ip}:443 (HTTPS/Traefik)")
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else:
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print(f" FAIL: TCP {server_ip}:443 (HTTPS/Traefik) unreachable")
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all_pass = False
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# WSS signaling
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if check_wss(livekit_domain):
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print(f" PASS: WSS {livekit_domain}:443 (LiveKit signaling)")
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else:
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print(f" FAIL: WSS {livekit_domain}:443 (LiveKit signaling) unreachable")
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all_pass = False
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# LiveKit TCP media port
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if check_tcp(server_ip, 7881):
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print(f" PASS: TCP {server_ip}:7881 (LiveKit media/TCP)")
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else:
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print(f" WARN: TCP {server_ip}:7881 (LiveKit media/TCP) unreachable")
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# LiveKit UDP media port
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if check_udp(server_ip, 7882):
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print(f" PASS: UDP {server_ip}:7882 (LiveKit media/UDP) send OK")
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else:
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print(f" WARN: UDP {server_ip}:7882 (LiveKit media/UDP) send failed")
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# TURN/UDP port — use STUN probe to verify the server actually responds
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if check_stun(server_ip, 443):
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print(f" PASS: UDP {server_ip}:443 (TURN/UDP) STUN response received")
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turn_available = True
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else:
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print(f" WARN: UDP {server_ip}:443 (TURN/UDP) no STUN response — TURN relay unavailable")
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if turn_available:
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print(" => TURN relay available — clients behind restrictive NATs can connect")
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else:
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print(" => TURN relay NOT available — only direct ICE connections will work")
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return all_pass, turn_available
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# ---------------------------------------------------------------------------
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# API helpers
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# ---------------------------------------------------------------------------
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def get_oidc_token(kc_url, realm, client_id, client_secret, username, password):
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"""Get an access token from Keycloak via direct access grant."""
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resp = requests.post(
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f"{kc_url}/realms/{realm}/protocol/openid-connect/token",
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data={
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"grant_type": "password",
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"client_id": client_id,
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"client_secret": client_secret,
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"username": username,
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"password": password,
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"scope": "openid email",
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},
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)
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resp.raise_for_status()
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return resp.json()["access_token"]
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def create_room(meet_url, token, room_name):
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"""Create a public room. Returns (room_id, livekit_config)."""
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resp = requests.post(
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f"{meet_url}/api/v1.0/rooms/",
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headers={"Authorization": f"Bearer {token}"},
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json={"name": room_name, "access_level": "public"},
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)
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if resp.status_code != 201:
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print(f" FAIL: Room creation returned HTTP {resp.status_code}: {resp.text}")
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sys.exit(1)
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data = resp.json()
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return data["id"], data["livekit"]
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def join_room(meet_url, token, room_id):
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"""Join an existing room. Returns livekit config."""
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resp = requests.get(
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f"{meet_url}/api/v1.0/rooms/{room_id}/",
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headers={"Authorization": f"Bearer {token}"},
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)
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resp.raise_for_status()
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return resp.json()["livekit"]
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def delete_room(meet_url, token, room_id):
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"""Delete a room."""
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requests.delete(
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f"{meet_url}/api/v1.0/rooms/{room_id}/",
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headers={"Authorization": f"Bearer {token}"},
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)
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# ---------------------------------------------------------------------------
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# WebRTC test
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# ---------------------------------------------------------------------------
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async def push_audio(source, stop_event):
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"""Push silent PCM frames until stopped."""
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frame = rtc.AudioFrame.create(SAMPLE_RATE, NUM_CHANNELS, SAMPLES_PER_CHANNEL)
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while not stop_event.is_set():
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await source.capture_frame(frame)
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async def run_webrtc_test(lk_url, token_a, token_b):
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"""
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Connect two participants to LiveKit. A publishes audio, B verifies reception.
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Returns (success, frames_received, publisher_identity, error_message).
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"""
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ws_url = lk_url.replace("https://", "wss://").replace("http://", "ws://")
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room_opts = rtc.RoomOptions(auto_subscribe=True)
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room_a = rtc.Room()
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room_b = rtc.Room()
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stop_audio = asyncio.Event()
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track_received = asyncio.Event()
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result = {"frames": 0, "publisher": ""}
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@room_b.on("track_subscribed")
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def on_track_subscribed(track, publication, participant):
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result["publisher"] = participant.identity
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async def read_frames():
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stream = rtc.AudioStream(track)
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async for _ in stream:
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result["frames"] += 1
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if result["frames"] >= 5:
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break
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await stream.aclose()
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track_received.set()
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asyncio.ensure_future(read_frames())
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try:
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# Participant A connects and publishes
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await room_a.connect(ws_url, token_a, options=room_opts)
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source = rtc.AudioSource(SAMPLE_RATE, NUM_CHANNELS)
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audio_track = rtc.LocalAudioTrack.create_audio_track("test-audio", source)
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options = rtc.TrackPublishOptions()
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options.source = rtc.TrackSource.SOURCE_MICROPHONE
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await room_a.local_participant.publish_track(audio_track, options)
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audio_task = asyncio.ensure_future(push_audio(source, stop_audio))
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await asyncio.sleep(1)
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# Participant B connects and waits for track
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await room_b.connect(ws_url, token_b, options=room_opts)
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await asyncio.wait_for(track_received.wait(), timeout=TIMEOUT)
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return True, result["frames"], result["publisher"], ""
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except asyncio.TimeoutError:
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return False, 0, "", "timed out waiting for audio frames"
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except rtc.ConnectError as e:
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return False, 0, "", f"LiveKit connection failed: {e}"
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except Exception as e:
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return False, 0, "", f"unexpected error: {e}"
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finally:
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stop_audio.set()
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try:
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await room_b.disconnect()
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except Exception:
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pass
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try:
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await room_a.disconnect()
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except Exception:
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pass
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# ---------------------------------------------------------------------------
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# Main
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# ---------------------------------------------------------------------------
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def main():
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creds = load_credentials()
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meet_domain = resolve_domain("lasuite-meet")
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# LiveKit signaling runs on its own subdomain (LIVEKIT_DOMAIN in the recipe
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# .env), which is `livekit-meet.<suffix>` — not a `livekit.` prefix on the
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# meet domain.
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from utils.tests.helpers import resolve_domain_suffix
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livekit_domain = f"livekit-meet.{resolve_domain_suffix()}"
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meet_url = f"https://{meet_domain}"
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kc_url = f"https://{resolve_domain('keycloak')}"
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room_name = f"webrtc-test-{int(time.time())}"
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print("Testing WebRTC media flow: publish audio, verify reception")
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print()
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# Preliminary checks
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print("Step 1: Connectivity checks ...")
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critical_pass, turn_available = run_connectivity_checks(meet_domain, livekit_domain)
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if not critical_pass:
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print()
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print("FAIL: Critical connectivity checks failed — cannot proceed with WebRTC test")
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sys.exit(1)
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# Authenticate
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print("Step 2: Authenticating both users ...")
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token1 = get_oidc_token(
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kc_url, creds["kc_realm"], creds["kc_client_id"], creds["kc_client_secret"],
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creds["kc_test_user"], creds["kc_test_pass"],
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)
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token2 = get_oidc_token(
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kc_url, creds["kc_realm"], creds["kc_client_id"], creds["kc_client_secret"],
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creds["kc_test_user2"], creds["kc_test_pass2"],
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)
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print(" PASS: Both users authenticated")
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# Create room
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print(f"Step 3: Creating room '{room_name}' ...")
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room_id, lk1 = create_room(meet_url, token1, room_name)
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lk2 = join_room(meet_url, token2, room_id)
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print(f" PASS: Room created (id={room_id})")
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print(f" PASS: LiveKit tokens obtained for both users")
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print(f" LiveKit URL: {lk1['url']}")
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# WebRTC media test
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print("Step 4: Testing WebRTC media flow ...")
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print(" Connecting participant A, publishing audio ...")
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print(" Connecting participant B, waiting for audio frames ...")
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success, frames, publisher, error = asyncio.run(
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run_webrtc_test(lk1["url"], lk1["token"], lk2["token"])
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)
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# Clean up
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print("Step 5: Cleaning up ...")
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delete_room(meet_url, token1, room_id)
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print(" Room deleted")
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# Result
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print()
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if success:
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print("PASS: WebRTC media flow verified")
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print(f" Audio frames received: {frames}")
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print(f" Publisher identity: {publisher}")
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if turn_available:
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print(" TURN was available — media may have used relay or direct ICE")
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print(" (check LiveKit logs for connectionType to confirm relay vs direct)")
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else:
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print(f"FAIL: WebRTC media flow test failed — {error}")
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if "timed out" in error or "connection" in error.lower():
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print()
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if not turn_available:
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print(" Hint: Neither direct ICE (7881/7882) nor TURN relay (UDP 443)")
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print(" are reachable. If the server has TURN enabled, verify UDP 443")
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print(" is published and not blocked by firewall.")
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else:
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print(" Hint: TURN relay port (UDP 443) is reachable but media still")
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print(" failed. Check LiveKit logs to verify TURN server started.")
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print(" Run: ssh <server> docker service logs <stack>_livekit | grep -i turn")
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print()
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print(" Try running from a machine with less restrictive network access,")
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print(" or verify TURN is enabled in the LiveKit config.")
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sys.exit(1)
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if __name__ == "__main__":
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main()
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