VoIPTechnik

VoIPTechnik refers to technology for transmitting voice communication over Internet Protocol networks, replacing traditional circuit-switched telephone networks for voice calls and related media.

It relies on packet-switching, where voice is encoded into data packets and sent over IP networks. Core elements include user endpoints (IP phones, softphones, mobile apps), call-control servers or PBX, and media gateways that connect to traditional PSTN networks. Signaling protocols coordinate call setup, teardown, and features.

The dominant signaling protocols are SIP (Session Initiation Protocol) and, to an extent, H.323. For media transport, RTP (Real-time Transport Protocol) carries audio streams, often with RTCP for monitoring. Codecs like G.711, G.729, G.726, G.722, Opus determine quality and bandwidth.

Networks require sufficient bandwidth, low latency, low jitter, and minimal packet loss. Quality of Service (QoS) and NAT traversal techniques (STUN, TURN, ICE) are common. Security aspects include encryption of signaling with TLS and media with SRTP.

Applications include enterprise UC systems, hosted Cloud PBX, IP-telephony for home offices, and contact centers. Benefits include reduced costs, scalability, remote work support, and feature-rich communication. Limitations involve dependence on internet connectivity and potential quality variations.

The field is evolving with cloud-based VoIP services (UCaaS), WebRTC-based communications, and integration with collaboration platforms; regulatory aspects such as emergency calling (E911) and lawful intercept considerations apply.