VoIP & Telecom Glossary
Comprehensive definitions for VoIP, SIP, cloud PBX, business messaging, and enterprise communications terminology.
A
A2P Messaging
Application-to-Person messaging where software systems send automated texts to consumers. Used for OTP codes, appointment reminders, and marketing campaigns. Regulated under TCPA and requires proper carrier registration (10DLC or short code) in the US.
ACD (Automatic Call Distributor)
A telephony system that automatically routes incoming calls to the most appropriate agent or department based on predefined rules. ACD systems use skills-based routing, priority queuing, and real-time availability data to minimize wait times and improve first-call resolution.
AI Agent
An autonomous software program that handles customer interactions using natural language processing and machine learning. VoIP-based AI agents can answer calls, qualify leads, book appointments, and escalate to human agents when a request exceeds their scope. Also called virtual agents or conversational AI.
ANI (Automatic Number Identification)
The telephone network's mechanism for transmitting the originating caller's phone number. ANI is distinct from Caller ID — it's a billing and routing signal that cannot be blocked by the end user, making it valuable for authentication and fraud prevention.
API (Application Programming Interface)
A set of protocols that allows software systems to communicate with each other. In telecom, APIs enable developers to add voice, SMS, and video capabilities to applications without building carrier infrastructure. CPaaS providers expose APIs for call control, messaging, and number management.
ATA (Analog Telephone Adapter)
A device that converts analog telephone signals into digital VoIP packets for transmission over IP networks. ATAs allow businesses to use existing analog phones and fax machines on a VoIP system without replacing hardware.
Auto Attendant
An automated telephone system that greets callers and routes them to the appropriate extension or department using touch-tone or voice prompts. Often called a virtual receptionist, it eliminates the need for a live operator and improves call handling efficiency.
Average Handle Time (AHT)
The average duration of a single customer interaction in a contact center, including talk time, hold time, and after-call wrap-up. AHT is a key call center KPI that balances operational efficiency with service quality — lower is not always better if it means rushed interactions.
Average Speed of Answer (ASA)
The average time callers wait in a queue before being connected to an agent. ASA is a critical call center metric that directly impacts customer satisfaction. Most service level agreements target an ASA of 20–30 seconds.
Account Code
A numeric or alphanumeric code entered by callers or agents to associate calls with specific clients, projects, or cost centers. Account codes simplify billing, call tracking, and reporting for professional services firms.
Agent (Contact Center)
A customer service representative who handles inbound or outbound calls, chats, or messages through a contact center platform. Agents are assigned queues, skills, and availability statuses within the ACD/CCaaS system.
Attended Transfer
A type of call transfer where the transferring party first speaks with the intended recipient before handing off the call. Unlike a blind transfer, this ensures the receiving party is available and briefed, improving the caller experience.
B
Bandwidth
The maximum rate of data transmission over a network connection, typically measured in Mbps or Kbps. VoIP calls require consistent bandwidth allocation — a G.711 call uses roughly 87 Kbps including overhead. Insufficient bandwidth causes audio degradation, choppy voice, or dropped calls.
BLF (Busy Lamp Field)
A visual indicator on an IP phone or operator console showing the real-time status (idle, busy, ringing) of other extensions. BLF enables receptionists and operators to manage call flow efficiently without calling or messaging colleagues.
Blind Transfer
A call transfer where the transferring party forwards the call to another extension or number without first speaking with the recipient. Also called an unattended transfer — the original caller is placed on hold while the transfer is completed, then connected whether or not the recipient answers.
Business SMS
Text messaging services used by organizations to communicate with customers, send alerts, and run marketing campaigns using business phone numbers. Business SMS requires compliance with TCPA and carrier registration (10DLC, short codes) to ensure deliverability and avoid filtering.
Bridged Line
A shared telephone line configuration where a single line appearance exists on multiple phones simultaneously. When one phone answers, others can monitor the call or pick it up. Common in executive/assistant setups and small offices.
B-Channel
In ISDN and PRI circuits, a 64 Kbps bearer channel that carries voice or data. A standard T1-based PRI has 23 B-channels plus one D-channel for signaling. B-channels represent the actual voice paths available for simultaneous calls.
BPO (Business Process Outsourcing)
The practice of contracting specific business operations — frequently customer service and call center functions — to third-party service providers. BPO contact centers rely heavily on VoIP, SIP trunking, and CCaaS platforms to serve clients across geographies.
Busy Lamp Field
See BLF. The visual indicator showing line and extension status across an IP phone system.
C
Call Center
A facility or virtual environment where agents handle large volumes of inbound or outbound telephone calls. Modern call centers have evolved into multi-channel contact centers supporting voice, email, chat, and SMS. Cloud-based call centers (CCaaS) allow agents to work from anywhere using softphones.
Call Parking
A PBX feature that places a call on hold at a specific extension number (a 'parking spot') so it can be retrieved from any phone in the system. Common in healthcare, hospitality, and retail where staff are mobile and cannot transfer calls directly.
Call Queue
A virtual waiting area where incoming calls are held until an agent becomes available. Queues play hold music, provide estimated wait times, and offer callback options. Queue management is a core function of ACD systems.
Call Recording
The capture and storage of voice calls for quality assurance, compliance, training, and dispute resolution purposes. Cloud PBX and CCaaS platforms offer on-demand or automatic recording with secure, searchable archives. Many industries have legal requirements for call recording disclosure.
Caller ID
The display of a caller's name and/or phone number on the recipient's phone before the call is answered. Caller ID uses CNAM database lookups and is distinct from ANI. Businesses can customize the outbound Caller ID shown to customers.
CCaaS (Contact Center as a Service)
A cloud-based contact center solution delivered as a subscription service. CCaaS platforms provide ACD, IVR, call recording, analytics, omnichannel routing, and workforce management without on-premises hardware investment. Major providers include Five9, Genesys, and Talkdesk.
CDR (Call Detail Record)
A data record that documents the attributes of a telephone call — including origination, destination, duration, start/end times, and disposition. CDRs are used for billing, compliance, and analytics. Also called Call Data Records.
CNAM (Caller Name)
The name field transmitted with Caller ID, stored in a national CNAM database and displayed on the recipient's phone. Businesses can register their company name to improve answer rates and brand recognition. CNAM delivery is a separate charge from basic Caller ID.
Codec
A software algorithm that compresses and decompresses audio data for transmission over IP networks. Common VoIP codecs include G.711 (uncompressed, highest quality, most bandwidth), G.729 (compressed, good quality, less bandwidth), and Opus (variable bitrate, ideal for WebRTC).
CPaaS (Communications Platform as a Service)
A cloud platform that provides communication APIs for voice, SMS, video, and messaging that developers embed into applications. CPaaS eliminates the need to build telecom infrastructure. Leading providers include Twilio, Vonage, and Bandwidth.
Conference Bridge
A system that connects multiple participants into a single audio call simultaneously. Conference bridges can be hosted (cloud-based dial-in numbers) or on-premises. Advanced bridges support recording, muting, screen sharing, and moderator controls.
Call Flow
The defined path and logic a call follows through a phone system from answer to resolution, including IVR menus, queue routing, transfers, and voicemail. Designing efficient call flows is critical to customer experience and operational efficiency.
Call Hunting
A PBX feature that rings a series of extensions or phones sequentially or simultaneously until someone answers. Also called ring groups or hunt groups. Ensures calls are answered even when the primary extension is unavailable.
Carrier
A licensed telecommunications company that provides underlying network infrastructure for voice and data services. In VoIP, carriers wholesale SIP trunks and DID numbers to resellers and businesses. Also called a telco or service provider.
CLI (Calling Line Identification)
The international term for Caller ID — the transmitted telephone number identifying the originating caller. CLI can be manipulated in VoIP systems, which has led to regulations like STIR/SHAKEN to prevent spoofing.
D
DID (Direct Inward Dialing)
A telephone service that allows a business to assign individual phone numbers to each employee or department that routes directly to their extension without an operator. DIDs enable large organizations to give each employee a unique, dialable number.
Dial Plan
The rules and patterns that define how a PBX interprets and routes dialed digits. Dial plans control internal extensions, external calls, emergency numbers, and feature codes. Proper dial plan design prevents toll fraud and ensures calls reach the correct destination.
DNIS (Dialed Number Identification Service)
A feature that delivers the number the caller dialed to the receiving business's phone system or contact center. DNIS enables a single trunk group to support multiple phone numbers with different routing logic based on which number was called.
DTMF (Dual-Tone Multi-Frequency)
The tones generated when pressing keys on a telephone keypad, each consisting of two simultaneous frequency tones. IVR systems use DTMF for menu navigation. VoIP transmits DTMF via RFC 2833 or SIP INFO to prevent audio codec compression from corrupting the tones.
Direct Routing
A Microsoft Teams feature that connects Teams to the PSTN via a certified Session Border Controller and SIP trunk, allowing businesses to use their existing carrier relationships. Direct Routing provides cost control and PSTN calling without Microsoft Calling Plans.
Digit Manipulation
The process of altering dialed or received phone number digits within a PBX or SBC to match required formats. Common uses include stripping leading zeros, adding country codes, or normalizing E.164 format for outbound routing.
DID Porting
See Number Porting. The transfer of a DID number from one carrier to another while retaining the same phone number.
Deskphone
A traditional IP-based telephone device designed for use at a desk or workstation. Modern deskphones support SIP, HD voice, color displays, and expansion modules for BLF monitoring. Also called a hardphone.
E
E.164
The ITU-T international telephone numbering standard that defines a globally unique format for phone numbers. E.164 numbers include a country code (e.g., +1 for North America) followed by the subscriber number, with a maximum of 15 digits. VoIP systems use E.164 for unambiguous international routing.
E911 (Enhanced 911)
The emergency calling system that automatically delivers the caller's location to emergency dispatch (PSAP) along with the callback number. VoIP providers are required by the FCC to provide E911 service, and businesses must keep location records accurate for each device or user.
Encryption (SRTP/TLS)
The protection of voice call content and signaling data from interception. SRTP (Secure Real-time Transport Protocol) encrypts the audio media, while TLS (Transport Layer Security) encrypts SIP signaling. Together they provide end-to-end call security.
Extension
A short internal phone number assigned to a specific user, device, or service within a PBX system. Extensions route internal calls without consuming external phone lines. Extension ranges are defined in the PBX dial plan.
Echo Cancellation
Digital signal processing technology that removes echo caused by audio feeding back into a microphone or from hybrid circuit reflection. Echo cancellation is built into IP phones and SBCs to ensure clear conversation quality, especially on long-distance or satellite calls.
Erlang
A unit of measurement for telecommunications traffic intensity, representing one hour of continuous call occupation on a single circuit. Erlang calculations are used to determine the number of trunks or agents needed to handle a given call volume at an acceptable service level.
Endpoint
Any device that originates or terminates a telephone call in a VoIP network. Endpoints include IP phones, softphones, ATAs, conference systems, and mobile apps. Each endpoint registers with a SIP server using credentials to receive calls.
F
Failover
Automatic switching to a backup system, route, or carrier when the primary fails. VoIP failover configurations use secondary SIP trunks, PSTN fallback, or geographic redundancy to maintain call service during outages. Failover SLAs are critical for business continuity.
Firewall Traversal
Techniques that allow VoIP traffic to pass through firewalls and NAT devices that would otherwise block or corrupt SIP and RTP packets. Solutions include STUN, TURN, ICE, SBCs, and Application Layer Gateways. Proper firewall configuration is essential for VoIP quality.
First Call Resolution (FCR)
The percentage of customer calls resolved during the initial contact without requiring a callback or follow-up. FCR is a primary contact center KPI because it directly correlates with customer satisfaction and operational cost. Industry average FCR is approximately 70–75%.
Full Duplex
A communication mode where both parties can transmit and receive audio simultaneously, like a normal phone conversation. Full duplex requires sufficient bandwidth and echo cancellation. Half-duplex systems (like walkie-talkies) only allow one party to speak at a time.
Fraud (Toll Fraud)
Unauthorized use of a business phone system to make expensive calls, often to international premium-rate numbers. VoIP systems are vulnerable to credential theft and PBX hacking. Prevention measures include strong passwords, IP whitelisting, call anomaly detection, and geographic call restrictions.
G
G.711
The ITU-T standard codec for PCM (Pulse Code Modulation) voice encoding, offering toll-quality audio at 64 Kbps per call. G.711 is the baseline VoIP codec and is used for PSTN interconnection. Two variants exist: G.711u (μ-law, North America/Japan) and G.711a (A-law, rest of world).
G.729
A compressed voice codec that delivers good quality audio at only 8 Kbps, making it ideal for bandwidth-constrained environments. G.729 uses predictive coding (ACELP) to achieve 8:1 compression versus G.711. It introduces slightly more latency and processing overhead than G.711.
Gateway
A device or service that connects different telecommunications networks or protocols. A VoIP gateway translates between analog/TDM telephone signals and IP/SIP packets. Gateways are used to connect legacy PBX systems, analog phones, and fax machines to VoIP infrastructure.
H
HD Voice (Wideband Audio)
High-definition voice calling that uses wideband audio codecs (such as G.722 or Opus) to deliver significantly richer, clearer sound than standard narrowband telephony. HD Voice captures frequencies up to 7 kHz versus 3.4 kHz for PSTN, reducing fatigue and improving intelligibility.
Hold Music (Music on Hold)
Audio content played to callers placed on hold. MoH improves the hold experience, reduces perceived wait time, and can deliver marketing messages. VoIP PBX systems support custom audio files per queue, ring group, or extension.
Hot Desking
A PBX feature allowing employees to log into any shared phone using their personal credentials, making the phone temporarily their own extension. Hot desking supports flexible and hybrid work environments where employees don't have dedicated desks.
Hunt Group
A configured set of phone lines or extensions that ring in a defined sequence or pattern (sequential, simultaneous, round-robin) when a specific number is called. Hunt groups maximize the chance a call is answered without the caller being sent to voicemail.
I
Intercom
A direct one-way or two-way audio communication system between specific phones on a PBX. Intercom calls typically auto-answer on the receiving phone, enabling immediate communication without the recipient needing to press a button. Used widely in manufacturing, schools, and healthcare.
IP Phone
A telephone device that transmits voice calls as digital packets over an IP network using SIP or other VoIP protocols. IP phones connect to Ethernet and register with a SIP server. They offer features unavailable on analog phones, such as HD voice, XML applications, and BLF.
IVR (Interactive Voice Response)
An automated telephony system that interacts with callers through voice prompts and touch-tone or speech recognition input. IVR systems route calls, collect information, and provide self-service options for tasks like account inquiries, appointment booking, and payment processing.
Intent Recognition
Natural language processing technology that identifies the purpose or goal behind a caller's spoken input. Used in AI-powered IVR and virtual agent systems, intent recognition allows callers to describe needs in natural language rather than navigating menus.
ISDN (Integrated Services Digital Network)
A legacy circuit-switched digital telephone network standard supporting voice and data on the same lines. ISDN is being decommissioned by carriers worldwide in favor of VoIP/SIP. PRI (Primary Rate Interface) and BRI (Basic Rate Interface) are ISDN variants still found in enterprise environments.
IP PBX
A Private Branch Exchange that uses Internet Protocol to route internal and external calls. IP PBX systems can be on-premises hardware, software running on servers, or hosted/cloud-based. They replace traditional TDM PBX systems with more flexible, feature-rich, and lower-cost alternatives.
ITN (Inverse Text Normalization)
The conversion of spoken words into their written form, particularly for numbers, dates, and units. In voice AI and ASR systems, ITN converts 'three one six four seven two' into '316-472' for readability and processing.
J
Jitter
The variation in delay between packets arriving over an IP network. In VoIP, jitter causes choppy or robotic-sounding audio when voice packets arrive out of order or unevenly spaced. Acceptable jitter for VoIP is generally below 30–50 milliseconds. Jitter buffers compensate for this variation.
Jitter Buffer
A temporary storage area that smooths out timing variations (jitter) in received VoIP packets before they are decoded into audio. Adaptive jitter buffers dynamically adjust their size based on network conditions. Too large a buffer increases latency; too small causes audio glitches.
K
KPI (Contact Center)
Key Performance Indicators are quantifiable metrics used to evaluate contact center performance. Core VoIP contact center KPIs include Average Handle Time, First Call Resolution, Average Speed of Answer, Abandonment Rate, Customer Satisfaction (CSAT), and Net Promoter Score (NPS).
L
Latency
The delay between a voice signal being sent and received in a VoIP call, measured in milliseconds. ITU-T G.114 recommends one-way latency below 150 ms for satisfactory voice quality. Latency above 400 ms causes noticeable conversation overlap and degraded experience.
LNP (Local Number Portability)
The regulatory requirement and technical capability allowing customers to keep their phone number when switching carriers. LNP applies to local, toll-free, and mobile numbers. The porting process typically takes 1–5 business days and requires coordination between old and new carriers.
Long Code
A standard 10-digit phone number (in the US/Canada) used for A2P or P2P SMS messaging. Long codes are designed for conversational messaging. High-volume A2P use requires 10DLC registration to prevent carrier filtering. Also called a virtual mobile number (VMN) in international markets.
Local Number
A telephone number with a specific area code geographic association that may or may not indicate the physical location of the subscriber. VoIP providers offer local numbers in thousands of area codes, allowing businesses to establish a local presence anywhere. Also called a geographic number or virtual number.
Loop-Back Test
A diagnostic procedure where a signal is sent from a device and looped back to verify transmission integrity. In VoIP, loopback tests identify echo, latency, and packet loss issues in the network path between endpoints.
M
Media Gateway
A network device that converts media streams between different telecom networks, such as from TDM circuit-switched networks to IP packet-switched networks. Media gateways are used at carrier interconnect points and to connect legacy equipment to VoIP infrastructure.
MMS (Multimedia Messaging Service)
An extension of SMS that supports the transmission of multimedia content — including images, audio, video, and GIFs — between mobile phones and messaging platforms. Business MMS requires carrier registration and is supported by major CPaaS providers.
MOS Score (Mean Opinion Score)
A standardized numerical measure of voice call quality on a scale of 1 to 5, where 5 is perfect and below 3.5 is generally considered poor quality. MOS scores are used to benchmark codec performance and monitor network quality for VoIP calls. G.711 achieves approximately 4.4 MOS.
MPLS (Multiprotocol Label Switching)
A high-performance private network routing technique that directs traffic based on short path labels rather than long network addresses. MPLS networks provide guaranteed bandwidth and QoS for voice traffic, making them popular for enterprise VoIP deployments with multiple sites.
Music on Hold
See Hold Music. Audio content delivered to callers placed in a queue or on hold within a PBX system.
Multi-Tenant PBX
A phone system architecture where a single instance serves multiple organizations (tenants) with complete logical separation of extensions, users, call routing, and billing. Multi-tenant PBX is the foundation of hosted VoIP and UCaaS provider platforms.
Mobile Twinning
A PBX feature that simultaneously rings both a desk phone and a mobile phone when a call arrives at an extension, allowing users to answer on either device. Mobile twinning ensures employees remain reachable without publishing their personal mobile numbers.
N
NAT Traversal
Techniques used to establish VoIP communications through Network Address Translation (NAT) devices, which would otherwise block incoming SIP and RTP traffic. Solutions include STUN, TURN, ICE protocols, and Session Border Controllers placed at network edges.
NLP (Natural Language Processing)
AI technology that enables computers to understand, interpret, and generate human language. In telecom, NLP powers virtual agents, conversational IVR, voicemail transcription, and sentiment analysis for contact center quality management.
Number Porting
The process of transferring a phone number from one telecommunications carrier to another while retaining the same number. Governed by FCC regulations (LNP), porting requires a Letter of Authorization (LOA) and verification of account details. Also called port-in (gaining carrier) or port-out (losing carrier).
NANP (North American Numbering Plan)
The integrated telephone numbering scheme covering the United States, Canada, and 20 other territories in the NANP. NANP numbers follow the +1 country code and NXX-NXX-XXXX format. The NANP currently includes over 800 area codes.
O
Omnichannel
A customer engagement approach that integrates all communication channels — voice, SMS, email, chat, social — into a unified platform with consistent context and history. True omnichannel systems share customer data across channels, allowing agents to switch between channels seamlessly.
Opt-In / Opt-Out
The consent mechanisms governing commercial messaging under TCPA and carrier guidelines. Opt-in is explicit written or verbal consent to receive messages; opt-out is the customer's request to stop receiving messages (typically via STOP keyword). Violations carry FCC fines up to $1,500 per message.
Origination
The inbound call service that delivers calls from the PSTN to a VoIP system. An origination provider receives calls placed to a DID number and routes them to the customer's SIP endpoint. Distinct from termination, which is outbound calling.
P
Packet Loss
The failure of data packets to reach their destination in an IP network, measured as a percentage of total packets sent. Packet loss above 1% causes noticeable audio quality degradation in VoIP calls — dropped words, silence gaps, or robotic sound. QoS configurations prioritize voice packets to minimize loss.
PBX (Private Branch Exchange)
A private telephone switching system used within a business to manage internal and external calls. Modern IP PBX systems connect to the PSTN via SIP trunks and support hundreds of features including voicemail, auto attendant, call recording, and conferencing.
PoE (Power over Ethernet)
A standard (IEEE 802.3af/at) that delivers electrical power to devices over Ethernet cabling, eliminating the need for separate power adapters. PoE switches power IP phones, wireless access points, and security cameras through standard Cat5e/Cat6 cables.
Presence
Real-time status information showing whether a user is available, busy, on a call, in a meeting, or offline. VoIP systems publish presence via SIP-based protocols (SIMPLE) or unified communications platforms, enabling smarter call routing and reducing unnecessary transfers.
PSTN (Public Switched Telephone Network)
The aggregate of the world's circuit-switched telephone networks operated by national, regional, and local telephone companies. VoIP calls connect to the PSTN through SIP trunks and gateways, enabling calls to and from any telephone globally.
PRI (Primary Rate Interface)
An ISDN standard interface that delivers 23 B-channels (voice/data) and 1 D-channel (signaling) over a T1 line in North America, providing 23 simultaneous calls. Many businesses are migrating from PRI to SIP trunking for flexibility and cost savings.
P2P Messaging (Person-to-Person)
SMS or MMS messaging between two individuals using standard mobile numbers, as opposed to A2P (application-to-person) messaging. P2P messaging is governed by different carrier rules than A2P and is not suitable for high-volume business messaging.
Provisioning
The process of configuring and activating devices, services, or users on a telecommunications platform. Zero-touch provisioning (ZTP) allows IP phones to auto-configure by fetching settings from a provisioning server based on their MAC address.
PSAP (Public Safety Answering Point)
A call center operated by government agencies responsible for answering 911 emergency calls and dispatching appropriate services. VoIP E911 services route calls to the appropriate PSAP based on the registered location of the calling device.
Q
QoS (Quality of Service)
Network configuration techniques that prioritize voice traffic over other data types to ensure consistent call quality. QoS uses DSCP (Differentiated Services Code Point) markings, traffic shaping, and bandwidth reservation. Without QoS, VoIP quality degrades when the network is congested.
Queue
A virtual waiting list where callers are held until an agent becomes available to assist them. Call queues can play hold music, provide position announcements, offer callback options, and overflow to voicemail. Queue management is central to contact center staffing and service level optimization.
R
Ring Group
A set of extensions that ring simultaneously or in sequence when a specific number or extension is called, distributing calls among a group of users. Ring groups differ from queues — they have no hold logic and ring all members at once (blast) or in a defined order.
Robocall Mitigation
Technologies and regulations designed to prevent unwanted automated telephone calls. The FCC's STIR/SHAKEN framework authenticates caller ID to combat spoofing. Businesses making legitimate automated calls must register with carriers and comply with TCPA consent requirements.
RTP (Real-time Transport Protocol)
The standard protocol for delivering audio and video over IP networks in real time. RTP packets carry the actual voice data in VoIP calls, typically transmitted over UDP for low latency. Each RTP stream is defined by codec, payload type, and sequence numbers for packet ordering.
RTCP (RTP Control Protocol)
A companion protocol to RTP that provides out-of-band statistics and control information for an RTP session. RTCP reports deliver packet loss, jitter, delay, and quality metrics used to monitor and troubleshoot VoIP call quality in real time.
Recording Announcement
A legal disclosure played to callers informing them that a call may be recorded, required in many jurisdictions under two-party consent laws. Recording announcements can be played automatically before connecting to agents or through IVR prompts.
S
SBC (Session Border Controller)
A specialized network device deployed at the border of VoIP networks to control signaling and media streams, providing security, interoperability, and QoS enforcement. SBCs protect against DoS attacks, normalize SIP signaling between different vendors, and manage NAT traversal.
Short Code
A 5- or 6-digit phone number used for high-volume A2P SMS messaging in the US and Canada. Carriers allow short codes the highest throughput of any number type, making them suitable for mass alerts, marketing campaigns, and time-sensitive notifications. On SIPSTACK, short codes are provisioned through sales and send at your plan's speed (see Flare sending speeds at /pricing/sending-speeds). Dedicated short codes require carrier approval and monthly fees.
SIP (Session Initiation Protocol)
The signaling protocol used to initiate, manage, and terminate multimedia communication sessions including voice calls, video calls, and messaging over IP networks. SIP is the foundation of modern VoIP and is defined in RFC 3261. It handles call setup, routing, and teardown.
SIP Trunk
A virtual phone line that uses SIP protocol to connect a business PBX to the PSTN over an internet connection. SIP trunks replace traditional PRI and analog phone lines, offering lower costs, greater flexibility, and instant scalability. Businesses can have unlimited concurrent calls with sufficient bandwidth.
SIP Proxy
A server that routes SIP requests between endpoints, performing address resolution and load balancing. SIP proxies are used by carriers and enterprise deployments to distribute call traffic across multiple servers and geographic regions.
Skills-Based Routing
An ACD routing method that matches incoming calls to agents based on their specific expertise, language, certifications, or other attributes. Skills-based routing improves first-call resolution and customer satisfaction by connecting callers with the most qualified available agent.
Softphone
A software application that enables voice calls over IP networks using a computer, smartphone, or tablet as the telephone. Softphones connect to a SIP server using the same credentials as a hardware IP phone. Popular among remote workers, they provide full PBX functionality without physical hardware.
SRTP (Secure Real-time Transport Protocol)
An encrypted version of RTP that protects voice media content from eavesdropping and tampering. SRTP uses AES encryption and HMAC-SHA1 authentication. Combined with TLS for SIP signaling, SRTP provides comprehensive end-to-end call encryption.
STIR/SHAKEN
A framework of interconnected standards (Secure Telephone Identity Revisited / Signature-based Handling of Asserted information using toKENs) mandated by the FCC to authenticate caller ID and combat spoofing. Carriers digitally sign calls with attestation levels (A/B/C) to indicate ID verification confidence.
STUN (Session Traversal Utilities for NAT)
A protocol that allows VoIP endpoints behind NAT to discover their public IP address and port mappings for establishing peer-to-peer media streams. STUN is lightweight but insufficient for symmetric NAT — TURN servers are used as a fallback.
T
TCR (The Campaign Registry)
The industry-managed registry for 10DLC SMS campaigns in the United States. Businesses must register their brand and campaigns with TCR before sending A2P SMS on long codes. TCR assigns trust scores that influence carrier throughput rates.
Termination
The outbound call service that carries VoIP calls from a business system to the PSTN. A termination provider routes the call from the customer's SIP infrastructure to the destination phone number worldwide. Termination rates vary by country and carrier.
TLS (Transport Layer Security)
The cryptographic protocol that provides secure, encrypted communication over networks. In VoIP, TLS encrypts SIP signaling to prevent call interception and tampering. SIP over TLS is often referred to as 'SIPS' and uses port 5061 by default.
Toll-Free Number
A telephone number with a distinctive three-digit prefix (800, 888, 877, 866, 855, 844, or 833 in North America) where the called party pays for the call rather than the caller. Toll-free numbers are widely used by businesses to encourage customer contact and project a national presence. VoIP providers offer virtual toll-free numbers with instant provisioning and flexible routing.
Transcription
The conversion of speech from a voice call or voicemail to text using automatic speech recognition (ASR). Transcriptions enable searchable call archives, voicemail-to-email delivery, real-time agent assistance, and compliance monitoring in contact centers.
Trunk
A communication channel or link that carries multiple simultaneous calls between telephone systems. SIP trunks deliver VoIP calls over IP networks; analog trunks use physical wiring; PRI trunks use ISDN standards. Trunk capacity determines how many simultaneous calls a system can handle.
TTS (Text-to-Speech)
Technology that converts written text into synthesized spoken audio. TTS is used in IVR systems, voicemail greetings, automated notifications, and AI voice agents. Modern neural TTS engines (like ElevenLabs and Microsoft Azure) produce highly natural-sounding voices.
TURN (Traversal Using Relays around NAT)
A protocol that relays media packets through a server when direct peer-to-peer communication (via STUN) is not possible due to restrictive NAT or firewalls. TURN ensures media connectivity in challenging network environments at the cost of server resource usage.
Two-Factor Authentication (2FA)
A security mechanism requiring users to provide two forms of identification before accessing a system. In VoIP admin portals and UCaaS platforms, 2FA typically combines a password with a TOTP code, preventing unauthorized access even if credentials are stolen.
10DLC (10-Digit Long Code)
A US industry initiative enabling businesses to use standard 10-digit phone numbers for high-throughput A2P SMS messaging with carrier approval. 10DLC requires brand and campaign registration with TCR and carriers set its throughput by the brand's trust score, positioning it between short codes and unregistered long codes. On SIPSTACK, a 10DLC number sends at the lower of your plan's speed and that carrier limit (see Flare sending speeds at /pricing/sending-speeds).
U
UCaaS (Unified Communications as a Service)
A cloud delivery model that integrates voice, video, messaging, conferencing, and collaboration tools into a single platform accessed over the internet. UCaaS eliminates the need for separate communication systems and on-premises PBX hardware. Examples include Microsoft Teams, RingCentral, and Zoom Phone.
Unified Communications (UC)
The integration of real-time and non-real-time communication services — including voice, video, instant messaging, presence, email, voicemail, and conferencing — into a cohesive user experience. UC enables seamless communication regardless of device, location, or communication channel.
Uptime SLA
A contractual commitment from a VoIP or cloud communications provider specifying the percentage of time their service will be operational. Enterprise-grade providers offer 99.999% uptime ('five nines'), meaning less than 5.26 minutes of downtime per year. SLAs typically specify credits for missed targets.
URI (Uniform Resource Identifier)
In SIP, a URI identifies a user or service endpoint, formatted as sip:user@domain or sips:user@domain for secure connections. SIP URIs serve as addresses for routing calls and messages to specific endpoints without requiring traditional phone numbers.
V
VLAN (Virtual Local Area Network)
A logical subdivision of a physical network that isolates traffic between different groups of devices. In VoIP deployments, separating voice traffic onto a dedicated VLAN ensures QoS, improves security, and simplifies troubleshooting by isolating voice packets from data traffic.
VoIP (Voice over Internet Protocol)
The technology that enables voice calls to be transmitted as digital data packets over IP networks instead of traditional circuit-switched phone lines. VoIP reduces communication costs, enables advanced features, and supports remote work. It underpins all modern cloud PBX and UCaaS systems.
Voicemail-to-Email
A PBX feature that automatically transcribes or attaches voicemail recordings as audio files and delivers them to a user's email inbox. Voicemail-to-email ensures messages are never missed and can be accessed from any device. Many systems also include AI-generated text transcriptions.
VoIP Gateway
A device that bridges VoIP networks and traditional PSTN/analog telephone infrastructure. VoIP gateways convert between SIP/RTP and TDM/analog signaling formats, allowing businesses to integrate legacy phone equipment with modern IP communications systems.
W
Wallboard
A real-time display board used in contact centers to show live metrics — such as queue length, agent availability, calls waiting, and service level — to supervisors and agents. Wallboards improve team awareness and allow real-time management decisions during peak call volumes.
WebRTC (Web Real-Time Communication)
An open standard and API that enables real-time audio, video, and data communication directly in web browsers without plugins or special software. WebRTC powers browser-based softphones, video conferencing, and live chat features on websites and SaaS applications.
Whisper
A PBX feature that allows a supervisor to speak privately to an agent during a live call without the customer hearing. Whisper coaching is used for real-time agent training and support. Related features include barge (supervisor joins the call three-way) and monitor (listen only).
Wrap-Up Time (ACW)
The period after a call ends during which an agent completes post-call tasks such as updating CRM records, logging notes, and selecting disposition codes before being available for the next call. Also called After Call Work (ACW). Minimizing wrap-up time increases agent efficiency.
X
XaaS (Anything as a Service)
An umbrella term encompassing cloud-delivered services delivered over the internet, including SaaS, PaaS, IaaS, UCaaS, CPaaS, and CCaaS. In telecom, the XaaS model has enabled businesses to access enterprise-grade communications without capital expenditure on hardware or software licenses.
Y
Yellow Pages / CNAM Database
Decentralized databases maintained by US carriers and data aggregators that store the registered name (CNAM) associated with telephone numbers. When you receive a call, your carrier queries a CNAM database to display the caller's name. Businesses can update their CNAM registration to display their company name on outbound calls.
Z
Zero-Touch Provisioning (ZTP)
An automated device configuration method where IP phones and other network equipment retrieve their configuration from a provisioning server upon first connection, requiring no manual setup. ZTP uses the device's MAC address to fetch the correct settings, dramatically reducing deployment time for large phone installations.