Today I will be talking about Voice over Internet Protocol (VoIP). I was vaguely aware of what VoIP technology was about before reading into it for this blog but a lot of this is new to me. I am going to do my best to explain what I have learnt, and why it is important. Let’s begin by looking at telephone networks!
I will begin by explaining what the Public Switched Telephone Network (PSTN) is. If you are familiar with computer networks, the PSTN is simply that – a network. Instead of being a network of computers and cables however, it is a network of switching centres and telephone lines. These switching centres are buildings that essentially connect calls together. It is key to briefly cover the PSTN since it is still very important and relied on for a lot of VoIP communication nowadays. Let’s talk about how companies conenct to the PSTN. Enter the Public Branch Exchange (PBX)!
A PBX is a private business phone system. You can think of it like a LAN but for telephones, whereas the PSTN is a WAN for telephones. A clarification that I need to make however is that PBX can refer to both the private business phone system as well as the PBX box, which is a physical device that is directly connected to the telephone lines. Standard PBX systems are connected to the telephone lines. Many people mistakenly call a VoIP server a PBX system, even though it connects to the Internet, and not to the PSTN.
VoIP and SIP
VoIP is technology used for voice communication over the Internet. Instead of having a traditional PBX system, we have a VoIP server (sometimes referred to as an IP-PBX). This VoIP server is then connected to the Internet.
Session Initiation Protocol (SIP) is a protocol that is used for initialising, maintaining, and ending calls made over the Internet. SIP trunking is a VoIP technology that allows a VoIP server to connect over the Internet. A SIP trunk is a connection from the VoIP server over the Internet. Think of it like a telephone cable but on the Internet.
If this voice traffic goes through the Internet and then needs to travel across the PSTN, it must be converted from digital data to analogue. This is achieved using VoIP gateways.
The word “codec” comes from the words “compression” and “decompression”. Codecs are a VoIP technology that are configured in the VoIP server that compress voice traffic data in order to have higher-quality calls and lower latency. Codecs are important because they reduce the bandwidth of calls but still maintain their quality. This is very efficient since the entire bandwidth won’t be used up by a call and can be used for other connections.
CISCO Meraki and SD-WAN
So now we know how amazing VoIP is for calling, I want to briefly mention a product that is commonly used in the world of VoIP. The CISCO Meraki router is a Unified Threat Management (UTM) device, meaning it is a single device that provides multiple security functions. Besides from providing security, CISCO Meraki routers have the ability to prioritise voice traffic due to its Quality of Service (QoS) feature.
I found this topic very confusing to read about. The information I found was not consistent and I spent a very long time gathering information and then struggling to piece it all together. The information in this article is correct to the best of my knowledge, however it is likely that I will have to add or correct certain things at a later date.
From what I have gathered, this is what an outgoing VoIP call looks like:
VoIP telephone —> Router —-> VoIP server (PBX) [–(SIP trunk)–> Internet] OR [—-> VoIP gateway —-> PSTN (switching centres/cell towers)]
Perhaps in the near-future, I will work on a hands-on project that showcases what I have learnt! I hope you enjoyed reading this short post and possibly have learnt something new. If you have anything to add, do let me know!