SIP TLS SRTP Explained: A Practical Guide to VoIP Encryption
📝 Blog Summary This guide breaks down why a working TLS handshake tells you nothing about whether your audio is […]
How to Stream FreeSWITCH Call Audio to an AI Model With mod_audio_fork?
📝 Blog Summary Connecting telephony channels to streaming speech models requires real-time media transport. With FreeSWITCH mod_audio_fork, developers attach a
VoIP Monitoring at Scale: MOS, QoS, and RTP Performance
📝 Blog Summary When a customer complains that a call sounded robotic, checking yesterday’s CDRs won’t tell you why. This
Speech-to-Speech vs STT-LLM-TTS | How to Architect Low-Latency SIP AI Voice Agents?
📝 Blog Summary Building a voice AI agent that works over a web browser is easy. Making it feel natural
How to Build Real-Time Agent Assist for Contact Centers?
📝 Blog Summary The most important aspect of real-time agent assist is decoupling live telephony media streams from asynchronous AI
RTPengine Explained | For Modern SIP & WebRTC Media Handling
📝 Blog Summary Standard SIP proxies handle text-based signaling but crash under heavy media loads. This guide explores how RTPengine
Kamailio vs FreeSWITCH: Pick Your Core – Routing or Media
📝 Blog Summary This blog breaks down Kamailio vs FreeSWITCH by role, not features, showing how SIP routing and media
SIP to PJSIP Migration Zero Downtime Strategy
📝 Blog Summary With Asterisk 21 removing chan_sip entirely, migration is no longer optional. However, moving from SIP to PJSIP
SIP Security for VoIP Providers: 7 OpenSIPS Modules That Matter
📝 Quick Summary Securing SIP isn’t about adding more tools, it’s about knowing where control actually belongs, and we’ve covered
What Are the Top Hosted PBX Security Strategies for 2026?
📝 Blog Summary Is your enterprise communication really secure? The blog explores how advanced PBX encryption, SBCs, firewalls, and predictive