Building a reliable and scalable VoIP backend is a task that requires a deep understanding of technologies and architectures. With remote work and online communication now the norm, this article will explain how to build a VoIP backend using WebRTC — one that ensures stable calls and is always ready to scale.
Why WebRTC Is the Backbone of Today’s VoIP?
WebRTC is an open-source protocol that enables audio and video calls directly through the browser, eliminating the need to download and install any additional software. It is ideally suited to develop scalable solutions due to the support of peer-to-peer connections and built-in security. The uniqueness of WebRTC lies in its use of standards supported by all major browsers, including Chrome, Firefox, and Safari. As a result, it provides a universal solution for developers.
WebRTC has several major strengths that enable it to be a perfect solution to create a scalable VoIP backend:
- Peer-to-peer connections: Reduce the server load by allowing media to be transmitted between members;
- Intrinsic security: The data are encrypted with the help of SRTP and DTLS protocols;
- Cross-platform compatibility: The modern browsers and devices, Android, and iOS are supported;
- Malleable signaling: WebRTC is agnostic of signaling protocol, and developers are free to supply fitting tools such as WebSocket or SIP.
All these qualities make WebRTC an all-powerful solution, but the development of a scalable solution requires a carefully planned architecture and choice of technology. In this respect, Acropolium has experienced more than 22 years of success in bespoke software development services with a high project success rate in the telecommunications sector in real-time solutions.
Architecture of a Scalable VoIP Backend
A VoIP backend is primarily composed of a signaling server, media server, STUN/TURN servers, and integration with third-party systems. Of each of them, let us descend to details.
i. Signaling Server: The Communication Heart
The signaling server is responsible for managing connections between clients. It exchanges metadata such as SDP (Session Description Protocol) and ICE candidates to establish a peer-to-peer connection. For scalability, the signaling server must be:
- Asynchronous: The support technologies of asynchronous operating, i.e., WebSocket, enable the connection of thousands of connections in parallel.
- Scalable: Supported on cloud vendors like AWS or Google Cloud with horizontal scaling;
- Reliable: Clustering and fault-tolerant mechanisms are used in order to reduce downtime.
For example, building a signaling server with Node.js and the Socket.IO library can support 10,000 simultaneous connections without sacrificing performance at all.
ii. Media Server: Stream Management
For peer-to-peer connection is not feasible for group calls, a media server is necessary. Two approaches are:
- SFU (Selective Forwarding Unit): Forwards media streams without mixing them, thus decreasing the load on the server. For the majority of modern applications, this is ideal;
- MCU (Multipoint Control Unit): Streams are mixed into a single one, which takes more resources but may be applied to complicated scenarios.
SFU is suited for more scalable solutions since it is less resource-hungry. As an example, the Janus media server can handle up to 50 participants in one video call with very little latency involved.
iii. STUN and TURN: How to bypass NAT
STUN (Session Traversal Utilities for NAT) and TURN (Traversal Using Relays around NAT) servers must be deployed to make WebRTC functional in the real-world networks. STUN is the protocol where devices are able to identify their public IP address, and TURN is the backup method in case they cannot connect to their direct IP address.
- STUN: Cheap and lightweight, sufficient for most cases;
- TURN: More expensive but necessary for 10–20% of connections, especially in corporate networks where firewalls are restrictive.
Coturn, as a TURN server, can ensure consistent application performance even in complex network configurations.
iv. Integration with SIP and PSTN
A gateway must be used to translate WebRTC signals to SIP to support connecting to legacy phone systems (PSTN) or VoIP networks. This is particularly significant with call centers or enterprise solutions. WebRTC can be integrated with FreeSWITCH so that employees can use regular phones right in the browser.
Performance and Quality Optimization
A VoIP backend that is scalable must ensure low latency and acceptable audio and video quality even under heavy load. Here are some best practices:
i. Adaptive Bitrate Management
WebRTC is available with intrinsic bitrate adjustment, which adapts the quality of video dynamically in response to the network environment. Monitor latency, packet loss, and jitter with the WebRTC API to adjust streams in real time.
ii. Codec Selection
For audio, we recommend the Opus codec for high quality at a low bitrate. For video, VP8 or H.264 are suitable, although in the year 2025, AV1 is gaining popularity due to better compression.
iii. Load Balancing
More than one media server can be installed with load balancing using cloud services such as AWS Elastic Load Balancer to balance traffic. This is especially important for high-user applications.
Integration with Business Processes
The VoIP backend needs to be integrated with existing business processes. This may include:
- CRM systems: Integration with Salesforce or HubSpot for automatic call logging;
- Analytics: A combination of solutions like Google Analytics to monitor user activity;
- Automation: Incoming calls can be processed by IVR or chatbots.
For example, a VoIP solution integrated with ERP may facilitate automation of customer interactions and improve order processing.
Conclusion: Your Path to Scalable VoIP
Building a scalable VoIP backend with WebRTC is no longer just a technical challenge—it’s a business necessity in today’s world of remote work and real-time communication. By combining a reliable signaling server, efficient media servers (SFU/MCU), and robust STUN/TURN infrastructure, you can ensure stable, high-quality calls for your users at scale.
Read More : History of VoIP and Internet Telephony: From The 1920s To Present Day
