12 Pain Points in Browser VoIP — and How to Fix Them
Building browser-based calling is powerful, but a handful of recurring issues create most of the user pain. Here’s a concise, field-tested playbook to diagnose and fix them fast.
Read more
SoftPage
Your No.1 Technology News Hub
Latest
Wi-Fi QoS for VoIP: DSCP & WMM Settings
Great microphones and codecs won’t save a call if your network treats real-time packets like background syncs. On Wi-Fi 6/6E, quality of service (QoS) is the difference between crisp audio and “can you hear me now?”. This guide explains how DSCP and WMM map voice traffic across wired and wireless hops—and the practical tweaks that actually move the needle.
Read more
DataChannel vs WebTransport vs WebSockets: When to Use Each
Real-time apps now have three strong browser options for moving data: WebRTC DataChannel, WebTransport, and WebSockets. They overlap, but they’re not interchangeable. Choosing the right one affects latency, reliability, back-pressure handling and deployment complexity.
Read more
SIP over WebSocket: Browser SIP Client for WebRTC PBXs
Most “click-to-call” demos end the moment a real PBX enters the picture. The missing piece is SIP over WebSocket (WSS): a standard that lets browsers speak SIP for signalling while WebRTC handles the encrypted media. The result is a true browser SIP client that can register, place calls, and integrate with your existing telephony—without installing anything.
Read more
WebRTC Topology: SFU vs MCU vs P2P
Most teams discover WebRTC through a simple peer-to-peer demo—and then hit a wall the moment real users, real networks, and real scale arrive. The hidden lever is your media topology: P2P, SFU, or MCU. Pick the wrong one and you’ll fight jitter, ballooning CPU bills, or fragile group calls. Pick the right one and you’ll deliver crisp audio/video with predictable costs.
Read more
WebRTC vs SIP: Understanding the Future of Internet Calling
When it comes to real-time communication, two acronyms dominate the conversation: WebRTC and SIP. Both play vital roles in enabling modern voice and video calling, yet they serve different purposes. For businesses and IT professionals, understanding the relationship between the two is essential for making the right technology choices.
Read more
How WebRTC SIP Clients Are Redefining Mobile VoIP
The way we make voice and video calls is undergoing a quiet revolution. Businesses and developers alike are moving away from heavy, app‑dependent VoIP systems and leaning into WebRTC SIP clients — lightweight, browser‑native solutions that blend the power of SIP protocols with the flexibility of modern web technology.
Read more
FreePBX WebRTC for Browser-Based VoIP
Open-source has always driven innovation in communications, and FreePBX remains one of the most widely used PBX platforms worldwide. As businesses embrace hybrid work, another technology is reshaping how teams connect: WebRTC.
Read more
WebRTC Asterisk: How to Power Browser-Based VoIP with Asterisk
For decades, Asterisk has been the open-source backbone of countless PBXs and VoIP systems. Meanwhile, WebRTC has emerged as the browser-native standard for secure, real-time communication. Put them together, and you unlock a powerful solution: WebRTC Asterisk integration, enabling direct browser-to-PBX calling without plugins or dedicated softphone apps.
Read more
Open Source WebRTC Softphones: The Developer Advantage
Open-source software has always been a driving force in communications — from Asterisk powering PBXs to FreeSWITCH enabling scalable VoIP. Now, the same collaborative spirit is fuelling a new frontier: the open-source WebRTC softphone.
Read more