Category: Telecom & VoIP - Page 2
Packet Loss Concealment (PLC) keeps VoIP calls clear despite network issues by intelligently filling in missing audio. Learn how G.711, Opus, and AI-based methods like WaveNetEQ hide packet loss-and which one works best for your needs.
Post-dial delay in VoIP can cause callers to hang up before a ring even starts. Learn how to tell if it's caused by signaling problems or media delays-and how to fix each one for better call success rates.
Fix your handset audio issues fast with this step-by-step guide for VoIP users. Learn how to troubleshoot silent speakers, dead microphones, and Bluetooth conflicts without replacing your device.
VoIP and landline systems handle Caller ID privacy very differently. VoIP offers advanced tools like encryption and AI screening, but requires setup. Landlines are simple but offer little control. Here’s how to choose what’s right for you.
Fix unreliable VoIP calls with this step-by-step port forwarding guide. Learn which ports to open, how to set up static IPs, and how to troubleshoot one-way audio and registration failures.
DTLS-SRTP is the modern, secure standard for VoIP media encryption, while SDES-SRTP is outdated and risky. Learn why DTLS-SRTP is mandatory in WebRTC and how to choose the right key exchange method in 2025.
Learn which codecs your IP phone supports in 2025 - from G.711 and G.729 to Opus and G.722. Get vendor-specific compatibility lists and real-world advice for optimizing call quality and bandwidth.
FXS and FXO ports let you connect analog phones and landlines to modern VoIP systems. Learn how they work, when to use each, and how to avoid common setup mistakes in 2025.
Compare G.711 and G.729 codecs for VoIP bandwidth usage. Learn which one saves bandwidth, which one sounds better, and how to choose based on your network, call volume, and budget.
SRTP adds less than 3% CPU overhead to VoIP calls and doesn't affect voice quality. Learn how encryption impacts codec performance, real-world numbers, and what systems still struggle with it.
Learn how dynamic and fixed jitter buffers affect VoIP call quality. Discover which one works best for remote teams, home offices, and enterprise networks based on real-world performance data and expert insights.
IPv6 eliminates NAT-related call drops and jitter in VoIP networks, offering direct end-to-end connections, better QoS, and faster call setup. While it uses slightly more bandwidth, the gains in reliability and quality make it essential for modern voice systems.
