SIP meaning: common terms and definitions

SIP meaning: common terms and definitions

Ever wondered about the origin or meaning of particular SIP terms, acronyms or buzzwords? We shed some light in this blog post.

Have you ever wondered about the origin or meaning of a particular acronym or buzz word in the world of telecommunications? In this blog post, we'll break down some common SIP terms and discuss their meaning in the VoIP world. We'll cover telephony data like CNAM or LRN, common metrics to measure performance like ASR, legal terms like LOA and everything in between.

What is the definition of SIP?

SIP is an acronym for Session Initiation Protocol and is a signaling protocol used for initiating, managing and ending communication sessions over the internet. It's most commonly used for Voice over Internet Protocol (VoIP) applications, such as video conferences and voice calls. SIP is based on the client-server model, meaning that a client initiates a request to the server, which then responds with an appropriate response.

The SIP protocol is formally defined in RFC 3261, published by the Internet Engineering Task Force (IETF). This specification describes SIP as an application-layer control protocol for creating, modifying, and terminating sessions with one or more participants.

The SIP protocol enables VoIP by providing a way for two parties to establish communication, exchange media and terminate the session. When a call is initiated, the client (or caller) sends a SIP request message to the server (or callee) with the details of the call (such as the caller's identity and the address of the callee). The server then responds with a SIP response message with details of the session (such as the accepted media types and the address of the caller).

SIP includes two types of messages known as SIP headers and SIP response codes. SIP headers are part of the SIP request and response messages and are used to convey additional information about the call, such as the caller's identity, the session's parameters, and the address of the callee. Response codes are also included in SIP messages and are used to indicate the status of the call, such as whether the call was accepted or rejected. The Internet Assigned Numbers Authority (IANA) maintains the official registry of SIP parameters, including header fields, response codes, and option tags.

By using SIP for VoIP, users can connect with each other over the internet and exchange media efficiently in real-time.

Understanding SIP, VoIP, and SIP trunking

Before diving into specific terminology, it's helpful to understand how SIP, VoIP, and SIP trunking relate to each other. While these terms are often used interchangeably, they serve distinct purposes in modern telecommunications.

SIP trunking is particularly valuable for businesses looking to build VoIP dialer solutions or modernize their communication infrastructure. According to the FCC, VoIP technology allows users to make voice calls using a broadband internet connection instead of a regular analog phone line, offering flexibility and cost advantages over traditional telephony.

Essential SIP and VoIP terminology

The following table provides quick reference definitions for the most common SIP and VoIP terms you'll encounter.

What SIP terms mean

Here are some of the most common SIP terms and with their definitions.

VoIP

Voice over Internet Protocol (VoIP) describes communication services such as voice, fax, SMS and voice messaging happen over public or private internet, rather than the public switched telephone network. This is also known as IP telephony. VoIP solutions are ideal for small businesses looking to reduce costs while gaining flexibility and scalability.

PSTN

The Public Switched Telephone Network is the world's analog, circuit-switched telephone network. It is also known as POTS (plain old telephone service) and the ISDN in some countries (Integrated Services Digital Network).

SIP

As explained above, SIP stands for Session initiation protocol and is the protocol that sets up the session between the individuals over the Internet. For a technical deep-dive into SIP implementation, see the Telnyx SIP Trunking Developer Documentation.

RTP

Real-time transport protocol is the protocol that actually sends the media when the session is established.

Soft phone

A soft phone is a software version of a phone that can be installed on your digital devices like laptops, smartphones, and tablets. They offer great flexibility when traveling and can be used as long as you're connected to the internet.

Hard phone

Hard phones are physical phones, as opposed to software-enabled phones. Hard phones can also be connected to the internet and use VoIP technology.

CLI

CLI stands for Calling Line Identification, which is the number associated with the person initiating the call.

CLD

CLD is an acronym for Calling Line Destination and is the destination number for a call or callee.

CNUM

CNUM describes the Caller ID Number, which is the telephone number of the calling party on an inbound call.

CNAM

Caller ID Name is often shortened to CNAM, and is the name associated with the CNUM of the calling party on an inbound call.

Attempted calls

The total number of calls made, whether they complete or not.

Connected calls

The number of calls attempted that return one of the SIP codes 200, 403, 408, 486, 487 or 480.

Completed calls

The number of calls attempted that return SIP code 200.

Duration

The length of the call, defined as from the INVITE to the BYE.

Billed duration

The billed length of the call, according to the billed intervals. For instance, if you're billed with 60/60 intervals, your billed duration will be rounded up to the nearest multiple of 60 in seconds.

ASR

ASR stands for Answer Seizure Ratio which is a common metric used to track call success rates. ASR is calculated as the percentage of attempted phone calls divided by the total completed calls.

ACD

In telephony, ACD is an acronym for Average Call Duration and defines the average length of the billed duration of completed calls.

SDC

Short Duration Calls (SDC) are calls that are less than 6 seconds in duration. This metric is commonly used to distinguish dialer traffic from conversational traffic.

NER

The network effectiveness ratio (NER) measures the ability of a network to deliver a call. This is calculated by dividing connected calls by attempted calls.

CDR

CDR stands for Call Detail Record. CDRs describe the characteristics of a call in table format. CDRs from Telnyx include the following: Caller (the initiating account), Original CLI (the initiating party's number), CLI (the number the call came from), Original CLD (the receiving party's number and prefix), CLD (the number that was dialed), Billing Prefix, Country, timestamps for start/answer/end, Duration, Billed Duration, Cost, Currency, Result (SIP response code), Remote IP, and Error Message.

TN

TN stands for telephone number.

ANI

Automatic number identification or ANI, is a feature that automatically determines the source telephone number on toll calls for billing purposes.

LRN

LRN is an acronym for local routing number. LRNs are unique 10-digit numbers formatted like a telephone numbers that are used for billing. It isn't a telephone number but represents an entire telephone switch where all the telephone numbers are routed. This allows for local number portability.

NPA

A number planning area (NPA) is a defined geographic area identified by a unique 3-digit code used in North America.

NXX

NXX is the middle part of a US number number that identifies the central office, otherwise known as the exchange, with the NPA in North America.

XXXX

XXXX represents the last 4 digits in the US number that identifies the station within the NXX in North America.

DID

Direct inward dial or DID numbers are telephone numbers allocated through one or more SIP trunk lines for a connection to a customer's PBX. All calls will be forwarded to these telephone numbers via the trunk.

PBX

PBX stands for private branch exchange and is a switching system that connects internal phones within a business and also connects them to the PSTN, VoIP providers and SIP trunks.

TFN

Toll-free numbers or TFNs, allow callers to reach businesses and individuals without being charged for the call. The called person is charged for subscribing to a toll-free number instead of the caller being charged for calling it.

SIP trunk

A virtual phone line provided by a SIP trunk provider like Telnyx. We use your data circuit, whatever that may be (T1, cable modem, DSL, Ethernet over copper), to connect your phone system back to our network.

FOC

In porting, FOC stands for firm order commit. This is an installation date for services usually used for number porting between service providers.

LOA

The Letter of authorization (LOA) is a document filled out by a customer when they switch telephone numbers that allows the new telecom provider to act on the customer's behalf. Used when a customer wants to keep their current telephone number or any other service which requires the transfer of information from one telecom provider to another.

What common SIP response codes mean

SIP response codes are three-digit numerical messages that contain information sent by the user agent server (UAS) to the user agent client (UAC). SIP response codes provide information about the status of the call. For a comprehensive guide to SIP methods, requests, and responses, see the Telnyx SIP Trunking Methods & Responses documentation.

INVITE

Indicates a client is being invited to participate in a call session.

ACK

Confirms that the client has received a final response to an INVITE request.

200

Indicates the request was successful.

180

An invite has been received by the user agent server (UAS), which is now attempting to alert the user.

183

Used to send session progress for a call that is still being set up.

BYE

Terminates a call and can be sent by either the caller or the callee.

Getting started with SIP trunking

Ready to implement SIP trunking for your organization? The Telnyx SIP Trunking Quickstart Guide walks you through creating a Mission Control account, configuring SIP connections, and making your first calls.

With Telnyx SIP trunking, users can count on carrier-grade quality and low latency communications. Our portal and APIs allow you to build your own scalable solution with highly configurable features for better control over all elements of your calls, including cost. Set up a plug and play VoIP system in minutes in the portal, or talk to an expert to learn more.