Version

SIP glossary

SIP (Session Initiation Protocol):

A protocol enabling the initiation, management, and termination of real-time sessions across IP networks, foundational for VoIP and unified communications.

VoIP (Voice over Internet Protocol):

A technology allowing voice and multimedia communications over Internet Protocol (IP) networks.

Remember, SIP is part of VoIP. VoIP is not just SIP.

UA (User Agent):

Software application located at the user's device that initiates and receives SIP requests, divided into User Agent Clients (UAC) and User Agent Servers (UAS).

SIP Registrar:

A server that accepts REGISTER requests from UAs to record their current location, used for routing SIP requests.

Proxy Server:

An intermediary entity that routes SIP requests based on policies and the requested user's location.

SBC (Session Border Controller):

A device controlling SIP signaling and media streams across network borders, ensuring secure and efficient communication.

SDP (Session Description Protocol):

Used within SIP to establish and manage communication sessions, describing session parameters like media format and transport protocols.

AoR (Address of Record):

A unique identifier within a SIP domain, used for SIP registration and routing calls/messages to the user's current location.

SIP URI:

A uniform resource identifier used to identify users or resources in SIP communications, specifying the user, domain, and transport parameters.

SIPS URI:

A URI scheme indicating secure communication sessions established using TLS, enhancing SIP's security.

TEL URI:

A URI scheme representing traditional telephone numbers within IP-based communications, facilitating interoperability with the PSTN.

GRUU (Globally Routable User Agent URI):

A permanent identifier for a SIP user agent, enabling direct routing of SIP messages to the specific user agent instance.

Wild Card URI:

A special URI used for bulk de-registration of contacts associated with an AoR, denoted by an asterisk (*) in the Contact header field.

Request Messages:

SIP messages sent to invoke actions, including INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INFO, REFER, MESSAGE, PRACK, and UPDATE.

Response Messages:

Server-sent messages indicating the outcome of a processed request, categorized into provisional, successful, redirection, client error, server error, and global failure responses.

Transactions:

The exchange of request and response messages between SIP entities, ensuring reliable communication. Transactions are divided into INVITE and Non-INVITE types.

Dialogs:

Peer-to-peer SIP relationships between two user agents, representing a session from setup to termination, including mid-session signaling.

RTP (Real-Time Protocol):

A protocol used for the real-time transport of audio and video over IP networks, crucial for SIP-based communications.

Codecs:

Algorithms for encoding/decoding media content for RTP transport, ranging from uncompressed formats like G.711 to compressed formats like G.729.

DTMF-Relay (Dual-Tone Multi-Frequency Relay):

A method for transmitting signaling information, such as DTMF tones, over RTP streams.

RTCP (Real-Time Control Protocol):

Provides feedback on the quality of media reception, complementing RTP by reporting on jitter, latency, packet loss, and round-trip time.

IMS (IP Multimedia Subsystem):

An architectural framework for delivering IP multimedia services, using SIP for session management across both wireless and fixed networks.

Start innovating with Mobius

What's next? Let's talk!

Mobius Software

As a company you'll get:

  • Get started quickly

  • Support any business model

  • Join millions of businesses

Questions? websupport@mobius.com