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VoIP & Telephony

Guides for running business phone systems: SIP and RTP firewall rules, codecs and bandwidth, call quality, PBX migration and number porting.

VoIP Call Quality: Jitter, Packet Loss and MOS Explained With 3 Targets

The three numbers that decide VoIP call quality (delay, jitter and packet loss), the targets to aim for, how MOS and R-factor are derived, and a repeatable way to measure a connection before and after changes.

September 30, 2026

VoIP One-Way Audio Fix: 6 Checks for NAT and RTP

A step-by-step way to find why one side of a call cannot hear the other: read the SDP, check NAT settings in PJSIP and 3CX, confirm RTP reaches the PBX, and remove SIP ALG and short UDP timeouts.

September 30, 2026

VoIP Codec Bandwidth: G.711 vs G.729 vs Opus, Per-Call Figures

How much bandwidth each VoIP call really uses once packet headers are added, how SRTP and VPN tunnels change it, and how to size a connection for a given number of concurrent calls.

September 30, 2026

Disable SIP ALG on 9 Routers and Firewalls (One-Way Audio Fix)

What SIP ALG does, why it breaks modern VoIP, how to confirm it is rewriting your packets, and the exact commands to turn it off on MikroTik, Cisco IOS, Juniper SRX, FortiGate, SonicWall and Linux.

September 30, 2026

SIP Ports Firewall Rules: 4 Setups for CSF, firewalld and pfSense

Which ports a PBX really needs (SIP 5060/5061 and the RTP media range), how to open them in CSF, firewalld and pfSense, and how to restrict SIP to your trunk provider so scanners cannot brute-force extensions.

September 30, 2026

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