Glossary

Telecom glossary: SIP, PSTN, SBC, E911, 10DLC, and other terms explained.

This glossary explains common telecom, SIP trunking, and business messaging terms in plain English so buyers, IT teams, and partners can make better decisions without drowning in jargon.

Definitions

Core telecom terms buyers and technical teams search for all the time.

Use the anchor links above to jump directly to a term. Each definition is written for clarity first, while still staying useful for more technical readers.

10DLC

10DLC stands for 10-digit long code and refers to application-to-person messaging sent over standard local phone numbers. It is commonly used by businesses for conversational texting, alerts, reminders, and other sanctioned A2P messaging use cases.

Most businesses using 10DLC in the United States need proper campaign registration and compliance with carrier rules around consent, content, and throughput. Without that registration, delivery can be limited or blocked.

A2P

A2P means application-to-person messaging. In practice, it describes messages sent by a software application, platform, or business system to an individual consumer or employee.

A2P traffic is governed differently from personal texting because carriers evaluate sender identity, consent, volume, and campaign type. Businesses using SMS at scale usually operate within A2P frameworks such as 10DLC, toll-free, or short code messaging.

CDR

CDR stands for call detail record. A CDR captures metadata about a call such as source number, destination number, time, duration, disposition, and sometimes routing path or billing data.

Telecom teams use CDRs to troubleshoot call issues, reconcile billing, investigate fraud, and understand calling patterns. They describe the call event, not the full audio content of the call itself.

CNAM

CNAM stands for Caller ID Name. It is the display name that can appear alongside a phone number when a call is delivered to the receiving party.

CNAM behavior can vary by carrier and endpoint, so it is not guaranteed in every call scenario. Businesses often use CNAM registration to improve recognition and answer rates on outbound calls.

Codec

A codec is the method used to encode and decode voice audio for transmission across a network. Common voice codecs affect sound quality, bandwidth use, and interoperability between systems.

Choosing the right codec matters in SIP environments because it can influence call clarity, network efficiency, and compatibility between phones, PBXs, SBCs, and carriers.

DID

DID stands for direct inward dialing. A DID is a phone number that routes inbound calls directly to a specific extension, user, department, or call flow inside a business phone system.

Businesses often maintain many DIDs across offices, teams, and campaigns. Managing DIDs well is important during migrations, number porting, and multi-location communications planning.

E911

E911 stands for enhanced 911. It refers to emergency calling capabilities that help route emergency calls and provide location information to emergency responders.

In modern business voice environments, E911 planning matters even more because employees may work across offices, homes, and mobile endpoints. Good E911 design aligns numbers, users, and location data intentionally.

FOC

FOC stands for firm order commitment. In number porting, it is the confirmed date and time when the losing and gaining carriers agree the port will complete.

An FOC is a key milestone in migration planning because internal teams often schedule testing, staffing, and communication around that cutover window.

LOA

LOA stands for letter of authorization. It is the document that gives a gaining carrier permission to port numbers or make service changes on behalf of the customer.

Accurate LOAs help prevent delays during number porting. Details like billing address, authorized entity name, and account information often need to match carrier records closely.

NAT

NAT stands for network address translation. It allows private IP addresses inside a local network to communicate with outside networks through a translated public address.

NAT can create problems for SIP and RTP traffic if not handled correctly because voice systems need to negotiate media paths and signaling details across network boundaries. SBCs and proper firewall design often help address this.

P2P

P2P means person-to-person messaging. It generally refers to messages sent by one person directly to another person, rather than automated traffic sent by an application.

The distinction between P2P and A2P matters because carriers regulate them differently. Businesses should be careful not to treat application-generated messaging as if it were personal texting.

PBX

PBX stands for private branch exchange. It is the business phone system that handles extensions, transfers, voicemail, ring groups, and internal call routing.

Traditional PBXs were often hardware appliances, while modern PBX environments may be software-based, cloud-based, or hybrid. The PBX sits between your users and the outside phone network.

POTS

POTS stands for plain old telephone service. It refers to traditional analog phone lines that many businesses historically used for voice, fax, alarms, or elevator phones.

POTS lines are increasingly expensive and harder to maintain, which is why many organizations are replacing them with SIP, wireless, or specialized modernization solutions.

PSTN

PSTN stands for public switched telephone network. It is the broader public phone network that allows calls to move between businesses, mobile devices, homes, and other carriers.

Even in cloud communications environments, the PSTN still matters because most business calls ultimately connect to or from this public network. Your carrier relationship determines how that path is handled.

QoS

QoS stands for quality of service. It refers to network policies that prioritize important traffic, such as voice, over less time-sensitive traffic like bulk downloads.

In voice deployments, QoS helps reduce jitter, delay, and packet loss that can damage call quality. It matters most in environments where voice shares bandwidth with other business applications.

RTP

RTP stands for real-time transport protocol. It is the protocol typically used to carry the actual voice audio stream during a VoIP call.

In a SIP call, SIP handles signaling and call setup, while RTP handles the media itself. Troubleshooting voice quality often involves looking closely at the RTP path.

SBC

SBC stands for session border controller. It is a device or software layer that secures and manages SIP traffic at the edge of a voice network.

SBCs are used for security, interoperability, policy enforcement, protocol mediation, and routing control. They are common in enterprise voice deployments and Direct Routing architectures.

SDP

SDP stands for session description protocol. It is used within SIP signaling to describe media details such as codecs, IP addresses, and ports for a call.

If SIP sets up the conversation, SDP helps the endpoints agree on how the media should flow. Voice troubleshooting often involves checking whether SDP negotiated the right path and format.

SIP

SIP stands for session initiation protocol. It is the signaling protocol used to establish, manage, and end voice or multimedia sessions over IP networks.

In business telecom, SIP is the foundation behind SIP trunking, many PBXs, and numerous cloud voice integrations. It handles call setup and control, not the actual audio stream itself.

SRTP

SRTP stands for secure real-time transport protocol. It is an encrypted form of RTP used to protect voice media streams during transmission.

Organizations use SRTP when they need stronger privacy and security for voice traffic, especially in enterprise, healthcare, legal, or compliance-sensitive environments.