SIP • Last Updated 9/29/2023

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.

By Sara Weichhand

SIP meaning and terms

Have you ever wondered about the origin or meaning of a particular acronym or buzz word in telephony? In this blog post, We're breaking down some common SIP terms and discussing 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 SIP?

Session Initiation Protocol (SIP) 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 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 also includes two types of messages known as SIP headers and SIP response codes. 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.

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

What common SIP terms mean

Here are some of the post common SIP terms, along with their definitions.

VoIP

Voice over Internet Protocol: Communication services such as voice, fax, SMS and voice messaging happen over the public internet, rather than the public switched telephone network. Also known as IP telephony.

PSTN

Public Switched Telephone Network: The world's analog, circuit-switched telephone network. Also known as POTS (plain old telephone service.

SIP

Session initiated protocol: This protocol sets up the session between the individuals over the Internet.

RTP

Real-time transport protocol: This protocol is what actually sends the media when the session is established.

Soft Phone

Software versions of phones that can be installed on your computer devices. They offer great flexibility when traveling and can be used as long as you’re connected to the internet.

Hard Phone

Physical phones, as opposed to software-enabled phones.

CLI

Calling Line Identification: The number associated with the person initiating the call.

CLD

Calling Line Destination: The destination number for a call or callee.

CNUM

Caller ID Number: The telephone number of the calling party on an inbound call.

CNAM

Caller ID Name: 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 SIP code 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

Answer Seizure Ratio: 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

Average Call Duration: The average length of the billed duration of completed calls.

SDC

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

NER

Network effectiveness ratio: Measures the ability of a network to deliver a call. This is calculated by dividing connected calls by attempted calls.

CDR

Call Detail Record: These records describe the characteristics of a call in table format. Telnyx's CDRs 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: The billing prefix for the number dialed
  • Country: The dialed number's country
  • start_time-stamp: Timestamp of the first INVITE
  • answer_time-stamp: Timestamp of the 200 OK
  • end_time-stamp: Timestamp of the BYE
  • Duration, sec: Total seconds of the call
  • Billed Duration, sec: Total billed seconds of the call
  • Cost: Total cost of call
  • Currency: The currency used for this call
  • Result: The SIP response code
  • Remote IP: The caller's remote IP
  • Error Message: The message that came with the SIP response code

TN

Telephone number

ANI

Automatic number identification: A feature that automatically determines the source telephone number on toll calls for billing purposes.

LRN

Local routing number: Used for billing and is a unique 10-digit number formatted like a telephone number. 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

Number planning area: A defined geographic area identified by a unique 3-digit code used in North America.

NXX

Part of the number that identifies the central office, otherwise known as the exchange, with the NPA in North America.

XXXX

Part of the number that identifies the station within the NXX in North America.

DID

Direct inward dial: 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

Private branch exchange: 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 number: 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

Firm order commit: An installation date for services, usually used for number porting between service providers.

LOA

Letter of authorization: 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.

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




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 with carrier-grade voice using Telnyx's elastic SIP trunking services, or talk to an expert to learn more.

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