30 Jun 2022
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SIP Trunking Terminology for Beginners

If you’re new to the world of SIP trunking, no doubt you’ve seen a lot of lingo that may not make sense at first. Well, we’ve gathered a list of key terms followed by a list of key initialisms and acronyms that will help you make sense of SIP trunking.


Voice over internet protocol, also known as IP telephony, is where communication services such as voice, fax, SMS, and voice messaging happen over the public internet rather than the public switched telephone network. Everything is moving through data cables and data centres instead of copper wire.


The most common and cost-effective mechanism to convert voice calls into VoIP calls that are sent over the SIP trunk is an IP PBX; that is, an internet protocol private branch exchange. An IP PBX is a private branch exchange (telephone switching system within an enterprise) that switches calls between VoIP users on local lines while allowing all users to share a certain number of external phone lines.

SIP Trunking

Telecommunications networks use connections that are referred to as “trunks” to connect us to the phone providers or carriers. SIP trunking uses a data connection to carry voice signals as VoIP to a service provider who can handle that voice signal. SIP trunking relies on internet protocols and services.

SIP stands for session initiation protocol, and SIP trunking is a service offered by an ISP.

The SIP protocol enables them to provision voice over IP (VoIP) connectivity between an on-premises phone system and the public switched telephone network (PSTN). Imagine a highway. Now imagine everyone’s voice and data travelling in separate cars. Is it possible for your voice and your data to carpool?

Voice data can be merged with other internet data, but we wouldn’t always recommend it, as we explained recently. If you prioritize voice interactions, it might be worth considering a dedicated internet access (DIA) circuit to complement your SIP trunking. Like peanut butter and chocolate, DIA and SIP are just better together.

Jitter, Latency, & Packet Loss

When an organization has not planned their IT infrastructure to ensure the network readiness required to adopt VoIP, the most common symptoms are jitter, latency, and packet loss.

Jitter refers to the variation in time that it takes each packet travelling over a network to reach its destination. In terms of SIP, jitter creates a poor audio experience.

Like jitter, latency is an audio quality problem in SIP trunking. It refers to the amount of time it takes a packet to reach its destination. When latency is too high, voice calls can be degraded to unacceptable levels.

Packet loss occurs when one or more packets of data fail to reach their destination. With VoIP calls, packet loss degrades call quality by dropping parts of the conversation, causing choppy audio and dropped calls.

Unified Communications (UC)

Unified communications is a concept describing the integration of enterprise communication services such as instant messaging, presence information, voice, mobility like forwarding your work calls to your mobile phone, etc. Basically, unified communications increases the capability to switch easily with various ways of communicating. The adoption of UC has been enabled and accelerated by the adoption of VoIP-based technologies. More recently, video calling became part of this. This can also include control, management, and integration of these communication channels.


An internet service provider (ISP) or internet telephony service provider (ITSP) could be your same old familiar phone service carrier or a carrier who specializes in VoIP Services as opposed to traditional landlines and circuit-switched voice services.


A key element of SIP trunking is the session border controller (SBC). A vital component of managing SIP traffic, the SBC is either a virtual or hardware device that acts as a form of gatekeeper between your private network and the public network service providers.

The SBC manages all kinds of essential functions and applications, such as transcoding, transrating, QoS, security, routing, and more.

Other Initialisms, Acronyms, & Key Terms


A device known as an analog telephony adapter, today it is a small device that converts an analog signal to a digital one. It is used to add analog devices such as legacy PBX or key systems, overhead pagers, or door alarms to a digital phone network, such as an SIP-based phone system.


Calling line destination. The destination number for a call, or the callee.


Calling line identification. The number associated with the person initiating the call, or the caller.


Caller ID name. This refers to the name associated with the CNUM of the calling party on an inbound call.


Caller ID number. This refers to the telephone number of the calling party on an inbound call.


For the SIP trunking world, a codec is software that compresses and decompresses voice signals over the network. The codec used by your SIP trunking vendor determines how much internet bandwidth you will need for each call.


Direct inward dialing (DID) is a service of a local phone company (or local exchange carrier) that provides a number or a block of telephone numbers for calling into a company’s private branch exchange (PBX) system. The format is a 10-digit phone number to individual people within a PBX system. A DID will allow a phone number to ring through directly to a specific phone at an organization instead of going to a menu where you are required to dial an extension to reach that number.

Hard Phone

Any classic handsets and headsets. These are physical phones.


A private branch exchange. That is, a telephone system within an enterprise that lets all users share external phone lines and also switches calls between users on local lines.


The predecessor to the PSTN is the POTS. That is, the plain old telephone service, employed using analog signal transmission over copper loops. No, really.


Public switched telephone network. This is the world’s circuit-switched telephone network.


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


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

Soft Phone

These are software versions of phones that can be installed on your computer devices. A benefit is that they offer great flexibility when travelling and can be used as long as you are connected to the internet.

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