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Pure-Python SIP Server & WebRTC Media Gateway

Real-Time Intercom & WebRTC
Communications Platform

Bridge physical SIP intercoms & facial authentication terminals (SAFR SCAN, Fanvil, Dahua, Hikvision, Axis) seamlessly with modern WebRTC browsers, guard stations, and access control systems. 100% on-premise privacy, sub-30ms latency, Opus 48kHz & G.711 audio, and remote door relay release.

RFC 3261 & RFC 7118 Compliant
Opus 48kHz HD Audio (< 30ms latency)
100% On-Premise Data Sovereignty
SIP Intercom Hub Web Dashboard
Server Online & Ready
SIP: 5060 | WSS: 8555 | HTTP: 8585
Live WebRTC Audio Bridge
DTLS-SRTP • aiortc Engine
Interoperable with Industry-Standard SIP Intercoms, Biometric Terminals & Softphones
SAFR SCAN
Fanvil Security
Dahua Technology
Hikvision Access
Grandstream
Axis Communications
Akuvox & 2N
Linphone Softphones
Interactive Experience

Experience the Hub in Action

Test real-time intercom calls, trigger DTMF door release commands, and monitor synthetic Opus audio spectrum directly inside this browser demo.

Intercom Directory 5 Online
Main Turnstile EXT 1000 • SAFR SCAN SC200
Main Entrance EXT 1001 • Fanvil i16SV
Loading Bay EXT 1002 • Dahua VTO2211D
Office Lobby EXT intercom1 • Hikvision DS
Perimeter Gate EXT intercom2 • Akuvox R29C
Live Camera Feed
STANDBY / IDLE
SIP / WebRTC + RTSP H.264
Main Turnstile (SAFR SCAN SC200-RF) • 192.168.1.110 • EXT 1000
IDLE
00:00
Microphone & Stream Level
DOOR SECURELY LOCKED

Active Dialog Diagnostics

Target Station Main Turnstile (SAFR SCAN SC200-RF)
Extension 1000
Audio Codec G.711 (8kHz) + WebRTC / H.264
Relay Contact DE-ENERGIZED (OPEN)
Signaling Roundtrip 18 ms
Modular Architecture

The Complete Real-Time Ecosystem

Much like Linphone's layered stack of softphones, liblinphone SDK, and Flexisip server, SIP Intercom Hub delivers a modular, tightly integrated communication suite tailored for physical access security.

SIP Server Core Engine

A high-performance pure-Python SIP registrar and routing proxy supporting standard UDP/TCP multi-transport on port 5060, RFC 3261 compliance, and MD5 Digest Authentication.

  • RFC 3261 & RFC 3264 Session Negotiation
  • RFC 2617 MD5 Digest User Authentication
  • Stateless / Stateful Dialog Proxying

WebRTC Media Bridge

Directly bridges traditional unencrypted UDP RTP media streams from hardware intercoms to end-to-end encrypted DTLS-SRTP WebRTC sessions in browsers using aiortc.

  • Opus 48kHz, PCMU (G.711u), PCMA (G.711a)
  • RFC 7118 SIP-over-WebSocket Signaling
  • Dynamic UDP Port Pool (10000 - 20000)

Web SIP Intercom Hub

A zero-install modern web application designed for security guards, front-desk concierges, and residential tenants. Features live camera viewing, interactive dialpad, and door releases.

  • Live Intercom Extension Directory & Status
  • Instant Door Relay Release (DTMF 1 / *)
  • Microphone Volume & Audio Codec Indicators

REST Management & IoT API

Integrate door phones into smart buildings, Building Management Systems (BMS), access cards, and CCTV automation using lightweight JSON REST endpoints.

  • POST /api/intercom/door-unlock
  • GET /api/registrations & /api/calls
  • Zero-Downtime Hot Reloading
Topology & Protocols

Engineered for Robust On-Premise Security

How SIP Intercom Hub orchestrates real-time audio/video streaming, NAT traversal, and door relays across heterogeneous network perimeters.

SIP Intercom Hub 3D Network Topology

End-to-End Signaling & Media Pipeline

Hardware intercoms communicate via standardized UDP SIP signaling, while web clients establish secure WebSocket tunnels. The Python core translates packets in sub-millisecond intervals without audio degradation.

1

Signaling Multiplexing (Port 5060 & 8555)

Simultaneous processing of RFC 3261 UDP datagrams from outdoor stations and RFC 7118 WebSockets from web browsers.

2

Bi-directional RTP / DTLS-SRTP Transcoding

The aiortc media engine converts unencrypted standard RTP packets into encrypted WebRTC frames with zero external ffmpeg process spawning.

3

Low-Latency DTMF & SIP INFO Door Unlocking

When the guard clicks unlock, the server injects standard RFC 2833 telephone-events or SIP INFO packets directly into the active RTP session.

Deployment Guides

Deploy in Minutes on Windows, Linux, or Docker

Choose between our native one-click Windows Service installer, isolated Docker Compose container, or bare-metal Python installation.

Windows Service Installation Wizard

The standalone Windows installer runs silently as a native background service (SIPServerService), automatically launches on boot, configures Windows Firewall, and serves the Web Hub.

Download Windows Installer v5.0.0 (39MB)
PowerShell (Admin) - Verify Service Status
# Check Windows Service Status
Get-Service -Name SIPServerService

# Open the Web SIP Console in your browser
Start-Process "http://localhost:8585"

Run with Docker Compose in 1 Command

Deploy in any Linux server, virtual machine, or local developer workstation with full network isolation:

docker-compose.yml
version: '3.8'
services:
  sip-intercom-hub:
    image: siphub/intercom-server:v5.0.0
    container_name: sip_intercom_hub
    restart: always
    network_mode: host # Optimal for RTP port range handling
    volumes:
      - ./config.yaml:/app/config.yaml:ro
    environment:
      - PYTHONUNBUFFERED=1
Terminal Command
docker-compose up -d --build

Pure Python Setup (Python 3.10+)

Run directly on bare-metal using standard pip dependencies:

Terminal / Bash
# 1. Clone repository
git clone https://github.com/siphub/sip-intercom-server.git
cd sip-intercom-server

# 2. Install requirements (aiortc, aiohttp, pyyaml, cryptography)
pip install -r requirements.txt

# 3. Launch SIP Server and WebRTC Media Gateway
python server/main.py

Trigger Door Release & Query Endpoints

Automate relays and check real-time intercom registration statuses from third-party software:

cURL - Remote Door Unlock Command
curl -X POST http://localhost:8585/api/intercom/door-unlock \
  -H "Content-Type: application/json" \
  -d '{"extension": "intercom1", "dtmf_key": "1"}'
JSON Response
{
  "status": "success",
  "extension": "intercom1",
  "action": "door_relayed",
  "dtmf_tone": "1",
  "relay_hold_seconds": 5
}
Hardware Integrations

Tested & Certified Intercom Hardware

Seamless interoperability with biometric access terminals, IP video door stations, and SIP softphones.

SAFR SCAN SC50, SC100, SC200-RF SAFR by RealNetworks
SAFR SCAN Terminals
SFR-SC50 • SFR-SC100 • SFR-SC200-RF

3D facial authentication & full-duplex SIP/WebRTC video intercom terminals with 1.3" & 3.5" capacitive touch displays, IP65/IK08 durability, OSDP/Wiegand, and multi-tech RFID/BLE.

SIP / WebRTC G.711 8kHz RTSP H.264/H.265 OSDP / Relay
Fanvil i16SV, i10V, i32V, i64 SIP Intercoms Fanvil Security
Fanvil IP Video Intercoms
i16SV • i10V • i32V • i64

Ruggedized IP65/IK10 outdoor SIP door phones featuring native Opus 48kHz HD audio, wide-angle 2MP night-vision camera, numeric access keypad, and dual dry-contact door relays.

Opus 48kHz RFC 3261 SIP H.264 Video DTMF Relay
Dahua VTO2211D-P-S2 & VTO3211D Video Intercom Dahua Technology
Dahua Villa & Gate Stations
VTO2211D-P-S2 • VTO3211D • VTO2202F

Slim PoE outdoor video intercom stations with 125° IR night vision, Mifare IC card reader, RFC 2833 DTMF lock release, and direct standard SIP registrar compatibility.

Standard SIP PCMU / Opus Mifare RFID RFC 2833
Hikvision DS-KV8113-WME1 & DS-KD8003 Modular Intercom Hikvision Access
Hikvision Modular & Villa Series
DS-KV8113-WME1 • DS-KD8003-IME1

Vandal-resistant IK08 stainless & aluminum door stations with True WDR 2MP fisheye cameras, backlit call buttons, standard SIP 2.0 firmware mode, and dual lock relays.

SIP 2.0 Mode G.711u / G.711a True WDR HD Dual Relay
Axis A8105-E, Akuvox R29C, 2N IP Verso Intercoms Axis • Akuvox • 2N
Axis, Akuvox & 2N Stations
Axis A8105-E • Akuvox R29C • 2N IP Verso

Enterprise network door controllers and multi-tenant touch panels supporting full-duplex Opus/G.711 audio, HTTP/REST webhooks, and RFC 3261 digest authentication.

Opus / G.711 HTTP API ONVIF / RTSP PoE+
Linphone Desktop & Mobile SIP Softphone Clients Softphone Clients
Linphone Desktop & Mobile
Windows • macOS • Linux • iOS • Android

Register open-source Linphone softphones alongside browser consoles to answer intercom calls, view H.264 video feeds, and send DTMF door release tones from any device.

UDP / TCP / TLS Opus 48kHz H.264 / VP8 RFC 2617 Auth
Competitive Advantage

How SIP Intercom Hub Compares

See why security teams prefer our focused, lightweight Python platform over complex legacy PBXs.

Feature Capability SIP Intercom Hub v5.0 Linphone (Flexisip) Asterisk / FreePBX Proprietary Cloud Intercoms
Native WebRTC Browser Client ✓ Included (Out of box) Requires Web SDK build Requires complex WSS configs Vendor App Only
Integrated Door Relay Control ✓ Dedicated DTMF & REST Generic DTMF dialpad Dialplan custom script Locked to proprietary hardware
On-Premise Privacy & Zero Cloud ✓ 100% Sovereign Data ✓ Open Source ✓ On-premise ✗ Mandatory Cloud Fee
Opus 48kHz Codec Priority ✓ Default Native ✓ Supported Requires extra module Low-bitrate G.711
Memory Footprint < 120 MB RAM ~ 250 MB RAM > 800 MB RAM Cloud Hosted
Windows Service Installer (.EXE) ✓ 1-Click Inno Setup Manual service setup Linux Only (WSL needed) Proprietary Agent
Licensing & Pricing

Transparent Enterprise Licensing

Straightforward, scalable on-premise licensing and professional engineering services tailored to your intercom and access control deployment.

BAS-LIC-HS Core Server

Base SIP Server License

$10,000.00 / license

Base SIP Server License for up to 50 intercom channels. Includes the core SIP registrar, WebRTC media bridge, web console, and first-year support.

CNL-LIC-ID Per Channel

Individual Channel License

$200.00 / channel

Individual Channel License for Audio and Video Devices (1 per channel). Enables full-duplex audio, video streaming, and remote door unlock per endpoint.

SUP-LIC-ID 1st Year Included

Reader Support & Updates

$30.00 / channel / yr

Reader Support and Update License - one per channel, per year. The first year is already included in the base license.

EPS-REM Professional Services

Engineering Daily Rate

$500.00 / day

Engineering Daily Rate for System Design, Sizing, Software Installation, and System Commissioning.

Download Center

Get Started Today

Download the production-ready package of your choice. Free for on-premise installation and evaluation.

Docker Deployment

Multi-Arch • Linux / macOS / Windows
  • Lightweight Alpine Python Container
  • Host Network Mode for Optimal RTP
  • Docker Compose Template Included
View Docker Guide

Source Repository

Python 3.10+ • Modular Micro-Stack
  • Complete Unit Test Suite (100% Pass)
  • Fully Customizable config.yaml
  • Extendable with Custom SIP Filters
Clone on GitHub
Support & Answers

Frequently Asked Questions

Everything you need to know about ports, security, video streaming, and hardware integration.

By default, SIP Intercom Hub requires:
  • Port 5060 (UDP & TCP): Standard SIP signaling for hardware intercoms
  • Port 8555 (TCP): WebSocket & Secure WebSocket (WSS) for browser clients
  • Port 8585 (TCP): Web Management Hub & REST API
  • Ports 10000 - 20000 (UDP): RTP audio and video media stream ranges
Yes! Because SIP Intercom Hub strictly implements RFC 3261 and RFC 2617 Digest Authentication, you can register any Linphone client (Android, iOS, Windows, macOS, Linux) directly using the credentials created in config.yaml.
When an authorized user clicks the door unlock button in the Web Hub or triggers the /api/intercom/door-unlock REST endpoint, the server transmits DTMF tones (RFC 2833 or SIP INFO) to the physical intercom station. The station energizes its physical dry contact relay, opening the electromagnetic lock or motorized gate.
No. The entire system is 100% self-hosted and runs on-premise on your local area network (LAN) or private VPN. No cloud subscriptions, no external tracking, and full privacy compliance (GDPR/HIPAA ready).