Tagged: WebRTC

Hand someone a browser and a headset and they’re on the phone — no install, no VPN, works from a hotel lobby. That’s the promise of WebRTC telephony, and the part that usually turns into a science project is everything around the codec: ICE, STUN/TURN, certificate handling, and the NAT scenarios that only fail on customer networks.

RustPBX treats browsers as first-class endpoints. Here’s what’s actually involved.

Your team can call each other internally. Now you need to connect to the PSTN through a carrier, let customers call in from web browsers, and manage it all without spinning up a separate SBC appliance.

RustPBX’s SBC addon handles this in a single runtime – no Janus, no FreeSWITCH, no extra media servers.

Every PBX starts with the same basic need: let people inside the organization call each other. Whether your team uses desk phones, softphones, or web browsers, it all comes down to configuring extensions and getting them registered.

This post walks through setting up internal VoIP communication in RustPBX from scratch, using its web console.

In the world of real-time communication, performance is paramount. WebRTC has become the standard for peer-to-peer audio, video, and data transfer. While the browser implementations are well-known, server-side and native implementations are crucial for building scalable infrastructure, gateways, and high-performance clients.

This article presents a performance benchmark comparing three prominent WebRTC implementations, with the code available at restsend/rustrtc:

1. RustRTC: A pure Rust implementation of WebRTC (the project under test).
2. webrtc-rs: The most popular Rust implementation (a port of Pion).
3. Pion: The industry-standard Go implementation of WebRTC.