SIP and VoIP glossary
Short definitions of the terms you meet in PBX panels, provider portals and SIP logs, each linked to a longer guide.
A
- Attended Transfer
- An attended transfer, also called a consultative transfer, places the first call on hold, lets the transferring party talk to the destination, then connects the two using a SIP REFER with a Replaces header.
- Auth Username
- The auth username is the name a SIP client sends in the Authorization header during digest authentication, which may differ from the user part of its SIP address.
B
- Blind Transfer
- A blind transfer, also called an unattended transfer, moves a connected call to a new destination without the transferring party first speaking to that destination, using a SIP REFER request.
C
- Codec (Audio Codec)
- A codec is the algorithm that compresses audio into a digital stream for transmission and decompresses it on reception; both ends of a call must agree on one.
D
- DID (Direct Inward Dialing)
- A DID is a public telephone number delivered over a trunk that the PBX routes directly to a particular extension, queue or other destination.
- DNS SRV Record
- A DNS SRV record, defined in RFC 2782 and applied to SIP in RFC 3263, maps a service and domain such as _sip._tls.example.com to one or more target hosts with ports, priorities and weights.
- DTMF (Dual-Tone Multi-Frequency)
- DTMF (dual-tone multi-frequency) is the keypad signaling method that represents each key as a pair of audio tones, used in VoIP to navigate menus, enter PINs and dial feature codes.
E
- Extension
- An extension is a short internal number configured on a PBX that identifies a user or device, and which a SIP client uses as its username when registering.
F
- Feature Code
- A feature code is a short dial string, usually prefixed with * or #, that a PBX interprets as a command to toggle a service such as call forwarding, Do Not Disturb or voicemail access.
G
- G.711 Codec
- G.711 is the ITU-T standard for 8 kHz telephone-quality audio at 64 kbit/s, available as ulaw (PCMU, North America and Japan) and alaw (PCMA, most other regions).
I
- ICE (Interactive Connectivity Establishment)
- ICE is the NAT traversal framework, defined in RFC 8445, in which both endpoints collect candidate addresses (local, STUN-reflexive and TURN-relayed), exchange them in SDP and test pairs to find a working media path.
J
- Jitter (VoIP)
- Jitter is the variation in the time between arriving RTP packets, measured as described in RFC 3550, which a receiver smooths with a jitter buffer at the cost of added delay.
N
- NAT (Network Address Translation)
- NAT is the router function that maps private internal IP addresses to one or more public addresses, which breaks SIP and RTP because both protocols embed addresses inside their messages.
O
- One-Way Audio
- One-way audio is a connected call in which RTP flows in only one direction, so one party hears the other but not vice versa, typically because NAT or a firewall prevents the return media stream.
- Opus Codec
- Opus is a royalty-free audio codec, defined in RFC 6716, that covers narrowband to fullband audio at bitrates from about 6 to 510 kbit/s and adapts to network conditions during a call.
- Outbound Proxy
- An outbound proxy is a SIP server to which a client sends every request, regardless of the Request-URI, typically to pass through NAT, firewalls or a provider's edge.
P
- Packet Loss
- Packet loss is the proportion of RTP packets sent that never arrive at the receiver, detected through sequence number gaps and reported in RTCP as described in RFC 3550.
- PBX (Private Branch Exchange)
- A PBX is a private telephone switching system that connects internal extensions to each other and to external lines, with an IP PBX doing so over SIP and RTP.
R
- Realm (SIP Authentication)
- The realm is a string sent by a SIP server in a digest authentication challenge that names the protection space and is included in the hash the client computes from its password.
- Registrar
- A registrar is the SIP server that accepts REGISTER requests and stores the binding between a user's address of record and the contact address where that user can currently be reached.
- RFC 2833 / RTP Events
- RFC 2833, superseded by RFC 4733, defines the telephone-event RTP payload that carries DTMF digits as structured packets inside the media stream instead of as audible tones.
- RTCP (RTP Control Protocol)
- RTCP is the control protocol paired with RTP, defined in RFC 3550, that exchanges periodic sender and receiver reports about packet loss, jitter and timing for a media stream.
- RTP (Real-time Transport Protocol)
- RTP is the protocol, defined in RFC 3550, that carries encoded audio and video in sequence-numbered, timestamped packets, usually over UDP, after SIP has set up the call.
S
- SDP (Session Description Protocol)
- SDP is a text format, defined in RFC 8866, carried in the body of SIP messages to describe the media streams of a session, including codecs, addresses, ports and security attributes.
- SIP (Session Initiation Protocol)
- SIP is the text-based signaling protocol, defined in RFC 3261, that locates endpoints and sets up, modifies and ends multimedia sessions such as voice calls.
- SIP ALG (Application Layer Gateway)
- A SIP ALG (Application Layer Gateway) is a router or firewall function that inspects SIP packets and rewrites the addresses inside them to assist NAT traversal, often incorrectly.
- SIP Proxy
- A SIP proxy is an intermediary server that receives SIP requests and forwards them toward the destination, applying routing rules and authentication without handling the media.
- SIP Trunk
- A SIP trunk is a service from a carrier that connects a PBX to the public telephone network over IP using SIP, carrying multiple concurrent calls in place of physical phone lines.
- SIP URI (SIP Address)
- A SIP URI is the address of a user or service in SIP, written as sip:user@host with optional port and parameters, or sips: when secure transport is required.
- Softphone
- A softphone is software that acts as a telephone on a general-purpose device, registering with a SIP server and using the device's microphone and speaker to place and receive calls.
- SRTP (Secure Real-time Transport Protocol)
- SRTP is the profile of RTP, defined in RFC 3711, that encrypts the media payload and adds authentication and replay protection to each packet.
- STUN (Session Traversal Utilities for NAT)
- STUN (Session Traversal Utilities for NAT) is a protocol, defined in RFC 8489, by which a client learns the public IP address and port that a NAT has assigned to it.
T
- TLS (Transport Layer Security)
- TLS is the cryptographic protocol, currently defined in RFC 8446, that encrypts and authenticates a TCP connection, used in SIP to protect signaling between a client and the next hop.
- TURN (Traversal Using Relays around NAT)
- TURN (Traversal Using Relays around NAT) is a protocol, defined in RFC 8656, in which a public relay server receives media from a client and forwards it to the peer when no direct path exists.
U
- User Agent
- A user agent is a SIP endpoint that both sends requests (user agent client) and receives them (user agent server), such as a softphone, desk phone or media gateway.
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