40 terms
VoIP glossary
Short, practical definitions of the terms you meet when buying and running SIP trunks and wholesale voice. Each entry starts with a one-sentence definition, then gives a real example and links to related terms.
Terms A to Z
A
- A–Z terminationA–Z termination is a wholesale voice service that lets you send calls to any destination in the world over a single interconnect, rather than buying routes country by country.
- ACDAverage Call DurationACD (Average Call Duration) is the total billed talk time on a route divided by the number of answered calls, usually expressed in minutes.
- ASRAnswer-Seizure RatioASR (Answer-Seizure Ratio) is the percentage of call attempts on a route that are answered, calculated as answered calls divided by total attempts.
B
C
- Caller ID (CLI)Calling Line IdentificationCaller ID, called CLI (Calling Line Identification) in much of the world, is the phone number presented to the person you are calling.
- CDRCall Detail RecordA CDR (Call Detail Record) is the log entry a switch writes for every call attempt, listing who called whom, when, for how long, on which route and with what result.
- CodecA codec is the algorithm that converts speech into digital packets and back again, trading audio quality against the bandwidth each call uses.
- Codec negotiation (SDP)Session Description ProtocolCodec negotiation is the exchange, carried in SDP (Session Description Protocol) inside SIP messages, where two endpoints agree which codec, IP address and port to use for a call’s audio.
D
- DIDDirect Inward DiallingA DID (Direct Inward Dialling) number is a phone number that routes incoming calls over the internet to your SIP trunk or PBX, without a physical line in that location.
- DTMFDual-Tone Multi-FrequencyDTMF (Dual-Tone Multi-Frequency) is the signalling used when someone presses a key on their phone, which IVRs, voicemail and payment systems use to read menu choices and PINs.
E
- E.164E.164 is the international standard for phone number format: a country code followed by the national number, up to 15 digits in total, with no leading zeros or trunk prefixes.
- ErlangAn Erlang is a unit of telephone traffic equal to one line busy continuously for one hour, used to work out how many channels a phone system needs.
F
G
- G.711G.711 is the standard uncompressed telephone codec, sampling speech at 8 kHz and sending it at 64 kbit/s, in two variants: µ-law (G.711u) in North America and Japan, and A-law (G.711a) almost everywhere else.
- G.729G.729 is a compressed voice codec that sends speech at 8 kbit/s, about one third of G.711’s bandwidth on the wire, at the cost of slightly lower quality.
I
- IP authenticationIP authentication is a way of securing a SIP trunk where the provider accepts calls only from a list of approved public IP addresses, with no username or password.
- IVRInteractive Voice ResponseAn IVR (Interactive Voice Response) is an automated phone menu that plays recorded prompts and routes callers based on the keys they press or the words they say.
J
L
- LatencyLatency is the one-way delay between someone speaking and the other person hearing them, measured in milliseconds.
- LCRLeast Cost RoutingLCR (Least Cost Routing) is the practice of automatically sending each call over the cheapest available route for its destination, usually with quality rules so cost never overrides a working call.
M
N
- NATNetwork Address TranslationNAT (Network Address Translation) is the router function that lets many devices with private IP addresses share one public address, and it is the most common cause of one-way audio on SIP calls.
- Non-CLI routeA non-CLI route is a voice termination route that does not guarantee your caller ID reaches the recipient, in exchange for a lower per-minute rate.
O
P
- Packet lossPacket loss is the share of voice packets that never arrive, heard as clipped words, gaps or a robotic sound on the call.
- PBXPrivate Branch ExchangeA PBX (Private Branch Exchange) is a business phone system that connects internal extensions, handles features like queues and voicemail, and links the office to outside lines.
- PDDPost-Dial DelayPDD (Post-Dial Delay) is the time between finishing dialling and hearing ringback or an answer, typically measured from the INVITE to the first 180 Ringing or 183 Session Progress.
- PRIPrimary Rate InterfaceA PRI (Primary Rate Interface) is a digital phone line that carries 23 voice channels in North America (T1) or 30 channels in Europe and most other regions (E1) over a dedicated circuit.
R
S
- SBCSession Border ControllerAn SBC (Session Border Controller) is a security and interoperability device that sits at the edge of a voice network, controlling which SIP traffic gets in and fixing incompatibilities between systems.
- SIPSession Initiation ProtocolSIP (Session Initiation Protocol) is the signalling protocol that sets up, changes and ends VoIP calls, using text messages similar in style to HTTP.
- SIP ALGApplication Layer GatewaySIP ALG is a router feature that tries to rewrite SIP messages as they pass through NAT, and it breaks VoIP calls so often that the standard advice is to switch it off.
- SIP trunkA SIP trunk is a virtual phone line delivered over the internet that connects your PBX or phone system to the public telephone network for inbound and outbound calls.
- SRTPSecure Real-time Transport ProtocolSRTP (Secure Real-time Transport Protocol) is the encrypted version of RTP, protecting call audio so it cannot be listened to if packets are intercepted.
- STIR/SHAKENSTIR/SHAKEN is the caller ID authentication framework used in the United States and Canada, in which the originating carrier digitally signs each call to confirm the caller is entitled to use the number shown.
T
- T.38T.38 is the protocol for sending fax over IP networks, converting fax signals into data packets rather than trying to carry them as audio.
- Tech prefixA tech prefix is a short string of digits added to the front of a dialled number so the provider knows which account, route or rate plan the call belongs to.
- TLSTransport Layer SecurityTLS (Transport Layer Security) encrypts SIP signalling, typically on port 5061, so phone numbers, credentials and call details cannot be read or changed in transit.
- Toll-free numberA toll-free number is a phone number that is free for the caller to dial, with the business that owns it paying for the inbound minutes instead.
W
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