Understanding the Infrastructure of WebRTC
WebRTC has revolutionised real-time communications, enabling audio, video, and data exchange directly in the browser. But for developers, IT admins, and telecom architects, the technology can feel like a maze of jargon: SIP, OpenSIPS, PBX, proxy server, endpoint-relay bridge, and so on.
In this article, we’ll demystify the core infrastructure behind WebRTC, showing where each piece fits and how they interconnect within a modern VoIP system.
🔗 What Is WebRTC?
WebRTC (Web Real-Time Communication) is an open-source framework that allows web browsers and mobile apps to support real-time communication via simple APIs — without requiring plugins or external software.
It handles:
- Peer-to-peer media streaming
- NAT traversal (via ICE, STUN, TURN)
- Codec negotiation and encryption
- Secure voice and video calling
But WebRTC rarely operates in isolation. It integrates with a broader VoIP ecosystem to communicate with SIP endpoints, legacy PBX systems, and cloud platforms.
Learn more about how WebRTC works.
📞 SIP and VoIP: The Protocol Layer
At the signalling layer, SIP (Session Initiation Protocol) is the backbone of most VoIP systems. It establishes, modifies, and terminates multimedia sessions like phone calls.
- VoIP (Voice over IP) is the umbrella term for delivering voice traffic over the internet.
- SIP is a protocol that enables VoIP communication.
- SIPS, the secure version of SIP, adds TLS encryption for call signalling.
For WebRTC-to-SIP communication to occur, you need a signalling gateway or SIP proxy — which brings us to OpenSIPS.
Read more on the difference between SIP and VoIP.
🧠 OpenSIPS: The Signalling Brain
OpenSIPS is an open-source SIP proxy server. It acts as an intelligent router for SIP messages and is often used to:
- Handle SIP registrations
- Route calls between softphones, SIP trunks, and PBX systems
- Enforce policies (e.g. NAT handling, authentication)
- Integrate with WebRTC proxies
In a WebRTC setup, OpenSIPS typically sits between the WebRTC client and the SIP back-end, translating browser-based signalling into SIP-compatible messages.
🔄 Proxy and Proxy Servers
A proxy server in the VoIP world is an intermediary that routes signalling and sometimes media traffic between clients and servers.
Roles include:
- Session management
- Codec negotiation
- Firewall traversal
- Load balancing
WebRTC often uses a dedicated WebRTC proxy or session border controller (SBC) to handle these tasks securely.
SIP proxy vs SBC – What’s the difference?
🖥️ PBX: The Legacy Core
A PBX (Private Branch Exchange) is a business phone system that manages internal calls and routes external ones via SIP trunks or VoIP providers.
In a WebRTC architecture, PBX systems:
- Serve as SIP endpoints
- Handle call control, voicemail, extensions, and conferencing
- Connect to WebRTC clients via a proxy and signalling bridge
Modern PBXs (like Asterisk or FreePBX) support WebRTC natively or via SIP bridges, allowing browsers and softphones to dial into the same telephony fabric.
How PBX systems connect to WebRTC
📲 Softphones and WebRTC Endpoints
A softphone is a software-based phone that runs on desktop or mobile, often leveraging WebRTC for media and SIP for signalling.
Softphones connect to:
- A proxy server for call setup
- A media server (like Jitsi or Kurento) for conferencing
- A PBX or SIP registrar for call routing
WebRTC endpoints can also be browsers, mobile apps, or embedded systems using JavaScript APIs and WebRTC libraries.
🔁 The Endpoint-Relay Bridge
The endpoint-relay bridge plays a crucial role in NAT traversal and media interworking. It ensures that:
- Media streams reach their destination even behind firewalls or NATs
- Different codecs or transport protocols are transcoded and relayed
- WebRTC clients can communicate with SIP or PSTN endpoints
Common relay tools:
- TURN servers (e.g. Coturn)
- Media bridges (e.g. FreeSWITCH or Janus Gateway)
🧩 Putting It All Together: A Simplified WebRTC Flow
- A user opens a softphone or browser to initiate a call.
- The WebRTC API gathers media and ICE candidates.
- Signalling is routed to OpenSIPS via a proxy server.
- The call reaches a PBX or another SIP endpoint.
- If needed, an endpoint-relay bridge handles NAT traversal and codec translation.
- Media flows peer-to-peer or via media servers as needed.
🛠️ Bonus Tools That Fit the Stack
- SBC (Session Border Controller) – often replaces or augments proxy functions
- STUN/TURN servers – handle NAT traversal
- SIP registrars – manage device authentication
- Media servers – enable multi-party calls, recording, and mixing
Siperb is an example of a modern WebRTC proxy that simplifies many of these layers by combining endpoint security, SIP compatibility, and browser-to-PBX routing in one encrypted platform.
✅ Conclusion
Understanding the infrastructure behind WebRTC is critical for designing reliable, secure, and scalable VoIP systems. Whether you’re working with SIP, OpenSIPS, PBX, or softphones, each component has a distinct role — and when they work together, they unlock powerful real-time communication.
We’d love your questions or comments on today’s topic!
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Thought for the day:
“We cannot change anything until we accept it. Condemnation does not liberate, it oppresses.”
Carl Jung
