Analog audio signals
Analog audio signals transmit voice data over telephone lines by modulating the frequency of sound waves to replicate pitch accurately. This same principle applies to radio wave transmissions.
ATA (Analog Telephone Adaptor)
An ATA is a hardware device that connects traditional telephones to the internet via a high-speed broadband connection. It converts analog voice signals into IP packets, supplies dial tone, and manages call setup.
Audio encoding
The ITU has standardized multiple audio codecs for H.323, all compatible with SIP (which is codec-agnostic). Key codecs include:
G.711 – 3 kHz audio encoded at 64 kbps, using PCM (the standard for traditional telephone networks).
G.722 – High-quality 7 kHz audio at 48, 56, or 64 kbps. Lower-bitrate variants include G.722.1 (24/32 kbps) and G.722.2 (~16 kbps).
G.723.1 – Optimized for low-bitrate speech compression (5.3–6.3 kbps).
G.728 – 3.4 kHz audio at 16 kbps, using small packet sizes (0.625 ms) to minimize delay.
G.729 – A modern 8 kbps codec with 15 ms packet sizes. Variants G.729 and G.729A differ only in mathematical implementation.
Speex – Open-source, patent-free, variable-bitrate (2.15–44.2 kbps).
GSM6.10 – Open-source at 13.3 kbps, though currently under patent dispute.
Audio Menu
A recorded phone menu offering verbal choices, commonly used in automated attendants, IVRs, and fax-on-demand systems. Users respond via voice commands or touch-tone inputs.
Audio Response Unit (ARU)
A computer telephony system using voice store-and-forward technology. Passive ARUs play pre-recorded messages, while interactive ARUs respond dynamically to caller input.
Audio Teleconferencing
Originally reliant on PBX conferencing circuits, early teleconferencing suffered from voice degradation and limited capacity. Dedicated conference bridges improved quality but required operator intervention. Modern PC-based systems allow self-managed conferences via touch-tone phones, email scheduling, or web browsers. Emerging solutions enable endpoints to handle mixing locally, eliminating centralized servers.
Bandwidth
The data transmission capacity of a communication channel, measured in bits per second (bps) for digital signals or Hertz (Hz) for analog. It can also refer to the range of frequencies a signal occupies.
Call Duration
The elapsed time from when a phone goes off-hook to when it is hung up after a call.
Circuit-Switched Networks
Used since 1878, these networks establish dedicated point-to-point connections for each call, limiting efficiency since no other traffic can share active switches.
Client (Softphone Client)
Software installed on a user’s device to enable internet-based calling.
Call Hunting
A feature that redirects inbound calls past busy signals or through multiple numbers until answered.
Class 5 (Telephony) Switch
A local telephone exchange providing dial tone, calling features, and digital services to subscribers. In SIP/VoIP/IMS networks, IP-based Class 5 switches (or Class 5 Application Servers) play a key role.
Clipping
The loss of speech segments, causing parts of words to drop out.
Cloud PBX
A hosted PBX service delivered over the internet (e.g., via AWS, Azure). Features:
Scalability: No on-premise hardware.
Remote management: Administered via web portals.
Integrates with SIP trunks, CRM tools, and UCaaS platforms.
Codec
Short for Compressor-Decompressor or enCOder/DECoder, a codec transforms data streams for transmission, storage, or encryption. Essential for videoconferencing and streaming media.
Compression
Reducing data size for efficient storage or transmission, particularly useful for large audio, video, and graphics files.
Conference Bridge
A device connecting multiple callers, with features like noise balancing, voice-activated muting, and touch-tone rejection. VoIP bridges use SIP/Megaco for control and RTP for media transport.
Data Compression
Shrinking files by identifying binary patterns, reducing bandwidth and storage needs. Effective compression can cut data to 40% of its original size (up to 90% for graphics).
Dial-Tone Delay
The lag (in milliseconds) between lifting a handset and hearing a dial tone.
DNS (Domain Name System)
Translates domain names into IP addresses. Recent additions like SRV/NAPTR and ENUM records support SIP/VoIP.
Dual-Tone Multifrequency (DTMF)
The touch-tone dialing system, assigning unique dual-frequency pairs to each key for easy detection.
E.164
The international numbering format for phone numbers (e.g., +1 555 123 4567). Critical for VoIP and PSTN interoperability.
Echo Cancellation
A feature to eliminate echo in voice calls caused by signal reflections. Critical for VoIP and audio conferencing.
Emergency 911 Calls
A North American emergency number routed to dispatchers for police, fire, or medical assistance.
ENUM (E.164 Number Mapping)
A DNS-based system that maps E.164 phone numbers to internet addresses (e.g., SIP URIs), enabling direct VoIP routing without PSTN gateways.
Fax over IP (FoIP)
Sending faxes over IP networks using protocols like T.38 to ensure reliability.
Find-Me/Follow-Me
A call-routing feature that rings multiple devices (e.g., cell, home, office) simultaneously.
Frame Relay
A packet-switching method using bandwidth on demand, capable of voice transmission with proper management.
Full Duplex
Simultaneous two-way communication without quality loss.
Gateway (VoIP)
Converts PSTN voice/fax calls to IP packets in real time.
H.323
An ITU standard for real-time voice/video conferencing over IP, supporting media gateways for packet conversion.
High-Availability
Designs ensuring constant system access, often using failover clusters.
Interactive Voice Response (IVR)
A phone-based system where callers interact via touch-tone or voice commands to query databases, hear synthesized responses, or update records.
Internet
The global network combining academic, government, and commercial subnetworks, serving as the backbone for the World Wide Web.
Internet Congestion
Slowdowns caused by excessive data on low-bandwidth or high-latency networks, leading to packet loss and degraded service.
Internet Telephony (IP Telephony)
Voice transmission over the internet, involving:
Client-side voice digitization.
Internet-based packet routing.
IP address lookup for connection.
PSTN gateways for phone-to-internet calls.
IP (Internet Protocol)
Defines data packet routing between sources and destinations.
IP Address
A unique identifier for devices on a network, either static or dynamically assigned.
IP Mapping
Geolocating IP addresses for network management.
IP Phone (VoIP/SIP Phone)
A device converting voice to digital packets for internet calls, supporting protocols like SIP/H.323 and codecs like G.711/H.261.
ISP (Internet Service Provider)
A business offering internet access, hosting, and related services.
ITU (International Telecommunication Union)
A UN-affiliated body setting global telecom standards.
Jitter
Signal timing fluctuations due to delayed/early packet arrivals.
Kbps (Kilobits per Second)
A data transfer rate metric (1 Kbps = 1,000 bits/sec).
Lag
Delays in data round-trip times, often from network or processing inefficiencies.
Latency
The delay between a data request and its fulfillment.
Mean Opinion Score (MOS)
A subjective 1–5 scale rating voice quality.
Messaging
Store-and-forward systems (e.g., voicemail, fax mail) for asynchronous communication.
NANP (North American Numbering Plan)
The system governing area codes and phone numbers in the U.S., Canada, and nearby regions.
NAT (Network Address Translation)
A method to map private IP addresses to a public IP (e.g., in home routers). Can cause issues for SIP/VoIP without STUN/TURN servers.
Opus Codec
A high-efficiency, low-latency audio codec for VoIP and streaming. Supports variable bitrates (6–510 kbps) and dynamic adjustment to network conditions.
Packet
A data unit containing payload, origin/destination details, and synchronization data, routed dynamically across networks.
Packet Loss
Data packets dropped due to congestion or errors.
Packet Switching
Efficient data transmission via routed packets, adaptable for voice/video on high-speed networks.
PBX (Private Branch Exchange)
A business phone system managing internal/external calls, evolving to support automated attendants.
Peer-to-Peer (P2P)
Decentralized networks where nodes act as both clients and servers.
POP (Point of Presence)
A carrier’s local network junction, or an email retrieval protocol.
POTS (Plain Old Telephone Service)
Basic analog phone service.
Protocol
Rules governing data transmission, including error handling and compression.
PSTN (Public Switched Telephone Network)
The global circuit-switched phone network.
QoS (Quality of Service)
Network performance metrics like delay and uptime.
Real Time
Minimally delayed communication (e.g., live calls).
Router
A device directing data packets between networks.
Sample Rate
Audio samples per second (Hz/kHz), determining bandwidth.
Service Provider
An entity offering and maintaining application services.
SIP (Session Initiation Protocol)
An IETF standard for interactive sessions (voice, video, messaging).
SIP Phone
A phone using SIP for internet calls, often with voicemail/blocking features.
SIP Trunking
A virtual alternative to traditional phone lines, using SIP to connect a PBX to the PSTN via the internet. Benefits:
Cost-effective: Eliminates physical PRI lines.
Flexible: Supports concurrent calls over a single connection.
Softphone
Software enabling calls via PCs/mobile devices with headsets.
Softswitch
Software replacing hardware switches at circuit-packet network junctions.
STUN/TURN/ICE
Protocols to help VoIP devices traverse NAT/firewalls:
STUN: Discovers a device’s public IP.
TURN: Relays media if direct connection fails.
ICE: Combines STUN/TURN for optimal routing.
TCP (Transmission Control Protocol)
A core internet protocol ensuring reliable data delivery.
Telephony
Telephony refers to the technology and systems used for voice communication over distances, traditionally via analog signals transmitted through physical networks like the PSTN (Public Switched Telephone Network)
UCaaS (Unified Communications as a Service)
Cloud-delivered services combining VoIP, video conferencing, messaging, and collaboration tools (e.g., Microsoft Teams, Zoom Phone).
Virtual Phone Number
A telephone number not directly tied to a physical line. Features include:
Cloud-based routing (e.g., forwarding to mobile, desk phone, or VoIP client).
Global availability (e.g., local numbers in foreign countries).
Used for IVR systems, call centers, and privacy protection.
VoIP (Voice over Internet Protocol)
Technology for transmitting voice calls over IP networks (e.g., the internet) instead of traditional PSTN. Key traits:
Uses codecs (e.g., G.711, Opus) to compress/decompress audio.
Relies on protocols like SIP, RTP, and WebRTC.
Enables features like video conferencing, instant messaging, and call analytics.
WebRTC (Web Real-Time Communication)
An open-source framework enabling browser-based voice/video calls and data sharing without plugins. Integrates with SIP and VoIP systems.
Wi-Fi (Wireless Fidelity)
A wireless networking technology that allows devices to connect to a local area network (LAN) using radio waves. Commonly used for VoIP calls, internet access, and data transfer without physical cables. Operates under IEEE 802.11 standards (e.g., Wi-Fi 6/802.11ax).