VoIP Phone System
I. VoIP Overview
Some businesses are hesitant to move from a traditional private brand exchange (PBX) with landlines to voice over internet protocol (VoIP) because of the lack of understanding of the technology.
It’s not that they have a hard time grasping it. There are just a lot of confusing information about VoIP on the internet. For example, the term VoIP is being used as a catch-all phrase for various communication technology, including the cloud, IP PBX, SIP, and other digital communication technologies. These terms may be related to VoIP but they are different concepts, altogether.
This VoIP technology primer attempts to bring clarity to the cloud-based technologies that offer digital collaboration.
This resource can help organizations understand what VoIP is, what it is not, and how it works.
II. What is VoIP?
VoIP stands for voice over internet protocol. As the name implies, the technology allows users to transmit voice communications over IP networks in real time. In general, VoIP is a group of telephony protocols that works through packet-switching technology where voice travels to its destination via individual network packets across various networks, including the internet. VoIP enables computer networks to function like traditional phone lines across the public switched telephone network (PSTN). The PSTN uses circuit-switched phone technology to transmit voice.
VoIP has only become a reliable business communications option in the last two decades, but the technology has been around for much longer.
Development of VoIP started around February 1995 when VocalTec Ltd., a company in Israel, came up with the idea to allow one user to call another through their computers with a microphone and speakers. This technology was initially limited. It only worked if the caller and the receiver had the same software setup.
The technology improved in the years that followed. These improvements gave rise to products like PC-to-phone and phone-to-phone solutions. VoIP was soon running across IP switches and routers. This made VoIP a standard feature on modern-day networking equipment.
VoIP has become a vital communications component of the modern business communications industry. Today’s communications solutions, including WaTech Alpha Voice, combine cloud voice solutions using VoIP and can interconnect with legacy phone systems through gateway connections to the PSTN.
III. VoIP communications vs. PSTN communications
To better understand the differences between VoIP and PSTN, here’s a side-by-side comparison.
Public Switched Telephone Network (PSTN)
Voice over Internet Protocol (VoIP)
Dedicated phone lines
When placing a phone call using PSTN, you are connecting over a circuit-switched path. The PSTN sets up a dedicated channel (or circuit) exclusively for the call. This connection remains for the duration of the phone call. Your voice will follow the same dedicated path during the call.
VoIP uses packet switching where the speaker’s voice is sampled and broken up into lots of little packets which are then transmitted over the internet. These packet payloads (your voice) are reassembled by the receiver’s phone and played by the phone handset.
Requires 64 Kbps per call
A single PSTN connection has a dedicated bandwidth of 64 Kbps. A T1 circuit with 23 channels can, therefore, handle 23 concurrent calls. Because of this limitation, the network isn’t able to handle additional capacity (calls) until someone hangs up to release network resources.
Ethernet requires 87 Kbps per call
VoIP codecs use just under 100 Kbps. The 87 Kbps includes the protocol stack header bytes. Both the high-quality G.711 codec and the high definition G.722 codec only use 87 Kbps in each direction.
No free calls, costly international calls
PSTN calls use the telephone carrier’s network infrastructure. So, in essence, every call made over PSTN is charged based on the use of this resource. Long distance and international calls are even more expensive since they are charged on a per minute basis or through a bundled minute subscription.
Free VoIP-to-VoIP calls, nominal subscription fees
You’re already paying a fixed fee for your internet connection (perhaps on a monthly basis), so you won’t incur any charges when you make a call using VoIP. Long distance calls, if not entirely free, have very low per minute rates and are often included in a regular monthly price.
Available at an extra cost
Call management features such as call waiting, call forwarding, caller ID, and call transfers are usually only available at an additional cost.
Standard and included
Most VoIP providers typically offer call management capabilities like call waiting, call forwarding, caller ID, and call transfers as standard or included features in their subscription plans.
Requires purchase of new hardware
Opening a new office (for instance) will require purchasing more hardware, on top of hooking it up to the original system, which can be costly and complicated. Upgrading your existing equipment is also quite expensive and requires a lot of time and energy.
Requires only bandwidth and software upgrades
Lines can easily be added as your needs grow. If one user moves to another state, the number and the phone can go with that person as well. If you need more capacity, contact your internet service provider (ISP) to buy more bandwidth.
Requires additional physical lines
Adding new extensions typically requires paying for the use of dedicated lines for each phone and paying a professional to punch down the wires for an additional phone and is therefore expensive.
Standard and included
If you need to add extensions to your phone system, all you have to do is to make the necessary configurations on your admin control panel and assign it to a user.
Remains active during outages
The PSTN service remains active even during power outages because the hardwired landline phones receive their electrical power from the PSTN’s central office (CO). Note—The base station of cordless phones requires a local electrical supply and will be unusable during a power outage.
No internet means no service
If power is out or the internet connection is down, your VoIP service may not work, especially without a backup power solution. The VoIP service is unavailable when internet connectivity is unavailable.
Location is traceable
Emergency service responders can instantly trace your location via the phone making the 911 call since the phone is associated with a known geographic location through its connection to the PSTN.
Location information may be untraceable or limited
Your e911 emergency calls cannot always be traced to a specific geographic location unless you, the subscriber, provide this information accurately to your VoIP provider.
IV. What VoIP is not
There are many misconceptions surrounding VoIP. This is especially true since the web uses the same term to refer to related but entirely different concepts.
Here are some common terms used interchangeably with VoIP, but are not the same thing:
—Private branch exchange (PBX), refers to the private phone system used by organizations to manage and route calls. A cloud PBX offers similar functionality, but it is hosted in the cloud instead of on-premises at the customer’s business. Cloud-based PBX systems interoperate with traditional phone lines like PSTN through gateways.
—Session initiation protocol (SIP) signals call initiation, changes, and terminations. Whereas SIP signals the phone calls, real-time transport protocol (RTP) carries the voice or media of the calls.
—The term has been used to describe both the technology (VoIP) and the hardware for it. For our purposes, IP phone will refer to the hardware or software device used to take and make VoIP calls. IP phones include hard phones and softphones.
- WebRTC—VoIP and WebRTC allow users to communicate from anywhere via the internet. WebRTC enables VoIP communication through a browser. WebRTC offers real-time communications (RTC) through simple application programming interfaces (API).
V. How VoIP works
A. Basic VoIP concepts
Before we get into how VoIP works, let us familiarize ourselves with some important concepts.
The primary technology components include:
—Voice and data are encapsulated within internet protocol (IP) packets. Source and destination IP addresses determine sender and receiver. IP packets are routed between different networks as they are passed from router to router across both private and public networks.
Encapsulation—When transmitting voice across a network, the voice is captured and segmented into small samples which are carried as payloads.
- The voice payload is a small portion of the audio stream. This sample is encapsulated by various network headers. The voice sample is the largest part of the message.
- A real-time transport protocol (RTP) header is attached to the front of the voice payload. Remember RTP carries media and SIP carries signaling.
- A transmission control protocol (TCP) header or a user datagram protocol (UDP) header—UDP carries the RTP header and then the media payload. TCP carries the SIP header for signaling.
- An internet protocol (IP) header holds the IP version, the source IP address, the destination IP address, packet size, and more.
Internet protocol (IP) address
—The network interface card (NIC) of every device to an IP network is given an IP address. IP version 4 addresses are built as four decimal numbers from zero to 255 that are separated by dots. For example, 192.168.1.1 is a commonly used private IPv4 address. IP addresses identify both the source and destination, the sender and the receiver, or the caller and the called party in a VoIP connection.
Port address—Port addresses range from zero to 65535. Just as IP addresses reference a specific network device, port addresses reference a listening service or program on a given device. An SIP server will listen for incoming requests on port address 5060 by default.
Transmission control protocol (TCP) vs user datagram protocol (UDP)
The most common types of VoIP protocol delivery stacks are TCP over IP or UDP over IP.
Transmission control protocol (TCP)
includes built-in handshaking before transmission and retransmission in case of error or message loss. Packets sent using TCP are guaranteed to be correct, and if it is lost it will be retransmitted. For these reasons, session initiation protocol (SIP) signaling is sent within a TCP message.
User datagram protocol (UDP)
is sometimes referred to as the “send and pray” method. UDP is a much simpler delivery method. UDP carries messages that do not need guaranteed delivery like voice or video. When voice or video messages are sent, the receiving party does not want to receive late messages. For these reasons, real-time transport protocol is sent within a UDP message.
TCP can easily cause delays if data packets get corrupted. UDP prioritizes speed over error protection and retransmission.
Session border controllers (SBC) act like firewalls to manage traffic by only allowing authorized subscribers to pass through. SBCs ensure high quality of service for voice calls. RingCentral uses SBCs between internet service providers (ISP) and RingCentral and between common carriers and RingCentral.
Real-time transport protocol (RTP)
RTP carries audio and video media streams with minimal delay. The RTP header contains information about the audio media file being streamed, including:
- Media content type
- Whether or not there have been ‘talk spurts’
- Sender identification
- Synchronization, time stamping, and sequencing data
- Loss detection
- Segmentation and reassembly rules
- Security (encryption)
- No retransmission capabilities (lost or delayed RTP messages are not resent)
The RTP header carries information about media streams and holds the payload format for the media type. RTP provides timestamps for synchronization and includes sequencing and time stamping. But, RTP offers no quality of service (QoS).
The RTP stream also has a companion protocol called RTCP or real-time transport control protocol, which is initiated and travels alongside the RTP stream. It does not carry media but generates RTP stream quality statistics.
Real-time Transport Control Protocol (RTCP)
RTCP provides feedback on the following:
- Quality of the media flow
- Number of lost packets
- Number of packets sent
- Number of bytes sent
- Jitter (difference in packet interarrival time) statistics
- Round-trip delays (the time it takes for a signal or message to get from the sender to the receiver and back)
RTCP provides valuable information when troubleshooting VoIP calls.
Each RTP stream is unidirectional. For both the caller and the callee to talk, you need a duplex stream. An RTP stream must be initiated in both directions.
Codecs, or coder-decoder, convert analog audio signals like your voice into digital samples. The received digital audio signal is converted back to an analog signal for the human ear upon arrival at the receiver’s phone.
Codec selection will determine the bandwidth required and the sound quality of the conversation.
Here are some of the most common codecs:
- G.711—Its formal name is pulse-code modulation (PCM). G.711 samples your voice 8,000 times per second and uses around 87 Kbps of bandwidth on an IP network. G.711 provides high-
quality voice end-to-end. The G.711 codec can be used on PSTN landline and VoIP packet samples with RTP.
- G.722—G.722 samples your voice 16,000 times per second and uses around 87 Kbps of bandwidth on an IP network. G.722 offers better voice quality and clarity than G.711 and is classified as HD voice.
- Opus—Opus offers a near-CD quality for voice with a bandwidth requirement between 60 and 80 Kbps. RingCentral uses Opus codec on their softphones.
- G.729—G.729 samples your voice at a rate of 8,000 Kbps and uses around 31 Kbps of bandwidth on an IP network. G.729 is not recommended for VoIP since it does not degrade well across IP networks.
Session Initialization Protocol (SIP)
SIP signals VoIP calls and is responsible for helping VoIP emulate telephone-like attributes. As said above, a lot of people mistakenly refer to VoIP as SIP as if they are the same. But as you can now see, VoIP is a group of protocols, and SIP is just one component working in the background to help VoIP calls work.
SIP is an open standard signaling protocol that can establish, manage, and terminate real-time communications over IP networks. SIP can be encapsulated or carried by either TCP or UDP.
SIP follows the client/server model, where the server refers to the VoIP provider (like WaTech Alpha Voice), and the client is the requesting phone.