EMZETT.Telephony
Image: Stevenrohner, CC BY-SA 4.0, Wikimedia Commons
In short: The transmission of speech over distance — an umbrella term for classic circuit-switched telephony and modern IP-based telephony (VoIP).
In more detail: Many companies have replaced classic phone-system technology with VoIP systems, which transmit speech like normal network data over the existing IP network — this saves separate cabling and simplifies integration with other software (e.g. video conferencing).
In Depth
From the exchange to a software service
The term “telephony” describes the entire process of voice transmission over distance — from signalling (call setup, dialling, ringing) through the actual voice transmission to call teardown. Historically, telephony was tightly bound to dedicated copper lines and mechanical, later electronic, exchanges: a call reserved a continuous physical connection between two parties for its entire duration, regardless of whether anyone was actually speaking or there was a pause in the conversation. This technique is called circuit switching and was the only available method for decades.
Increasing digitisation gradually shifted these functions into software. The first step was ISDN (Integrated Services Digital Network) in the 1980s/90s: speech was already digitally encoded, but still transmitted in a circuit-switched way — a fixed bandwidth was reserved per channel. The real turning point came with VoIP (Voice over IP), which transmits speech entirely packet-switched over normal IP networks, exactly like any other data traffic.
Signalling and media stream kept separate
An important concept of modern telephony is the separation of signalling and media stream. Signalling (usually via the SIP protocol, Session Initiation Protocol) handles the “surroundings” of a call: who’s calling whom, is the call accepted or rejected, which codecs are negotiated, when does it hang up. The actual audio stream, by contrast, runs over a separate protocol (usually RTP, Real-time Transport Protocol) directly between the endpoints or via a media server. This separation allows additional features like call forwarding, hold queues or conference calls to be implemented purely in the signalling layer, without touching the audio path itself.
Consolidation in companies
For companies, switching to IP telephony usually also means tangible cost savings and simplification: instead of a separate phone system (often a proprietary hardware box from one vendor), separate cabling (classic phone lines alongside the network cable) and separate maintenance contracts, everything runs over the same network infrastructure as the rest of the traffic. A softphone (a telephony app on a PC or smartphone) replaces the physical phone, and features like automatic call forwarding, voicemail-to-email transcription, or direct integration into chat/video-conferencing tools (clicking a contact directly starts a call) become pure software features instead of additional hardware.
Quality and typical problems
Voice quality with VoIP depends heavily on network quality: packet loss shows up as brief dropouts or crackling sounds, high and especially fluctuating latency (jitter) as unpleasant delay or choppy speech. For business-critical telephony, Quality of Service (QoS) is therefore often configured on the network, prioritising voice packets over less time-critical traffic (e.g. file downloads).
See also: VoIP, Telephones