VoIP Features

VoIP Technical Glossary - Protocols You Should Know

Written by Emily Boucher
June 7, 2023
7 Mins Read

What is SIP protocol

A SIP (Session Initiation Protocol) protocol initiates, maintains, and terminates sessions with one or more participants in a peer-to-peer protocol. The protocol applies to voice calls, instant  messaging, and other realtime communications. 

SIP protocol originated in internet technology, rather than telecommunications. It was initially developed by Mark Handley, Henning Schulzrinne, Eve Schooler, and Jonathan Rosenberg in 1996. While the original use of the application-layer protocol was to support phone calls, it quickly found uses for all real-time calling and messaging. 

SIP is transported over two subtypes of porting protocols:


TCP (Transmission Control Protocol) transports digital data packets in a slower, more reliable manner. It is typically used for email, file transfer protocol, and web browsing.  


UDPSIP trunk, Centrex and PBX (User Datagram Protocol) transports digital data packets faster, but is less secure. It is typically used for streaming, gaming, and VoIP.

SIP uses a process similar to the HTTP request/response model to send and receive data. When initiating a call (handshake), the SIP will open a UDP/TCP port. Throughout a voice call (or other realtime communication), the client or server will send a ‘ping’ to the recipient at regular time intervals, requiring a ‘pong’ be sent back by the recipient as soon as possible in order to maintain the connection.

What is VoIP Codecs

VoIP codecs, short for ‘encoder-decoder’ is the term for the software algorithms that compress and decompress voice signals into digital data. They are necessary for enabling voice communication over internet-based networks and systems.  

Different VoIP codecs use different algorithms for compression and decompression, depending on the use case (audio, video, voice, and text), affecting the quality and performance of the voice call. 

Examples of some commonly used VoIP include: 

G.711 (bandwidth 64 kbps): Precise speech transmission, high bandwidth requirements. One of the earliest codecs. Most likely to be used for LAN (Local Area Networks), rather than the internet.

G.722 ((bandwidth 48/56/64 kbps): A more adaptive follow-up to G.722, capturing a greater ranger of frequencies for improved call quality/clarity. 

G.723.1 ((bandwidth 5.3/6.3 kbps): Highly compressed audio to retain quality of sound. Usable with dial-up, other low-bandwidth environments but requires more processor power. 

G.729 (bandwidth 8 kbps): A licensed, error-tolerant, low-bandwidth codec purchased indirectly through the hardware (phones, gateways) using it.

GSM (bandwidth 13 kbps): Highly compressed, free to use for many hardware/software platforms. Used in older GSM cellphones (updates have been made since this was introduced).

iLBC (bandwidth 15 kbps): Internet Low Bit Rate Codec; free, open source, robust against data packet loss. Used by many VoIP apps.

Speex (bandwidth 2.15 / 44 kbps): Uses a variable bit rate to minimize bandwidth usage. Preferred for many VoIP apps.

SILK (bandwidth 6 to 40 kbps): Developed by Skype, then licensed as open-source freeware. SILK is used as a base for a newer codec called Opus, which is preferred by WhatsApp. 

What is Voiceover IP 

VoIP stands for Voiceover Internet Protocol, and is the blanket term for any realtime call made over the internet, rather than traditional telephone lines. The protocol translates analog voice signals into digital data packets that can be transmitted via the internet. VoIP calls can be made using a computer with a broadband connection, or with a VoIP-enabled phone.

Using VoIP services may enable a user to bypass a broadband connection or a traditional telephone while accessing additional features like email, messaging, file sharing, and video conferencing. However, VoIP service may not work during power outages, nor provide access to white page listings or emergency services.

What is SIP environment (Proxy / Registrar / SBC)

A SIP environment is a system or network that uses the Session Initiation Protocol to enable communication over the internet. It allows different devices and systems to communicate with each other for voice calls, video communication, instant messaging and file sharing.

Session Initiation is carried out between a registrar and a proxy server. A SIP registrar allows web and software applications to associate a username with a network address, much like a digital phone book. This registration is sent to the proxy server to show that the client exists. The SIP proxy server then serves at the entry into the larger server network. When the proxy server receives client information from the registrar, it can initiate and route a session with a recipient across the larger network, creating the connection. 

Once the connection is made, a device called a Session Border Controller (SBC) acts to manage communication sessions between networks and users. At the start of a session, the SBC works to ensure the session remains secure, and that all users are authorized to be on the call. In addition, the SBC manages quality control via bandwidth management, call routing, and managing interoperability between vendors. Stemming from VoIP networks, SBC technology regulates all real-time communications that enable VoIP, video, text chat, and other realtime collaboration.

What is the difference among SIP trunk, Centrex and PBX?


A Centrex system (“central exchange”) is a type of telecommunications system offered to businesses and government agencies, where the equipment is owned and maintained by the service provider. Centrex negates the need for the client to purchase and maintain an in-house telephone switch (PBX, or public exchange)

Centrex is offered by telephone companies as a way for businesses and government agencies to get access to a full set of telephone system features without having to purchase and maintain an in-house telephone switch (called a PBX, or Private Branch eXchange). Centrex was attractive to organizations that wanted a minimal capital outlay and that needed ongoing system maintenance included in a predictable monthly price. This telecom option is popular with businesses that have multiple locations, a distributed workforce, and require a unified communications system. 

This type of system offers features such as station-to-station dialing, consultation hold, three-way calling, call waiting, call transfer, and call forwarding.

SIP Trunking

A SIP Trunk system, on the other hand, uses a IP-enabled PBX (functionally, a network of virtual phone lines) to make a receive calls to anyone with a phone number. A SIP trunk can integrate with a business phone system to connect to the public switched telephone network (PSTN). Whatsapp is a commonly-used application that leverages SIP trunking for faster and more reliable call setup. SIP Trunking is often seen as a more flexible and scalable solution for businesses that wish to bypass phone costs. This technology is also what enables the flexibility of virtual phone numbers.


PBX (Private Branch eXchange) is a type of internal telephone network where a central service provides voice, data and other communications services at a single location. This allows for shared lines among multiple users. PBX is typically associated with analog telecom services, although SIP trunking allows digital services to be integrated into the system.


PSTN (public switched telephone network) is a networked group of interconnected public telephone systems used for government and business phone systems. At present, PTSN networks are almost entirely digital save for last-mile telephone office connections to users. 


Broadworks is an enterprise-grade calling and collaboration platform operated and maintained centrally by Cisco and partner providers. As a telecommunications system, it enables service providers to offer, set up, and maintain a range of high-performance communications tools which include voice calls, video, messaging and collaboration tools to customers. These services are maintained and configured for customers on the provider’s own network infrastructure

While Broadworks services place a strong emphasis on telecom offerings such as call-forwarding, voicemail, and conferencing, this system integrates with Cisco WebEx to provide a full range of real-time communications solutions.


FreeSWITCH is an internet-focused open-source platform, and is used to build PBX, videoconferencing services and other communications services. 

FreeSWITCH is distinct from internet-based platforms such as Zoom because it utilizes SIP trunking and web solutions to increase compatibility across a wide range of use cases, devices, and systems. This scalability and flexibility make it popular for building for custom communications solutions. Twilio is one popular example of a FreeSWITCH-enabled service platform.

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So, download the app today and get started learning more about how our system works. Textr Apps are available on Android, iOS, and web browsers.

About the Writer

Written by Emily Boucher

Copywriter who codes. Why? Because these languages are writing, I write for all audience and computers.