Buzz
Video calls? Now available too. Chad Springer / Getty ImagesMain Points
- VoIP (Voice over Internet Protocol) is a system that enables voice communication through the internet by transforming analog audio signals into digital data.
- It provides a cost-effective and versatile solution compared to conventional phone systems, often including additional features like call forwarding, voicemail, and conference calls at no extra charge.
- VoIP can be used in various forms — via analog telephone adapters, IP phones, or directly through software on a computer — with each method requiring an internet connection to work.
If VoIP is a new concept to you, prepare to revolutionize your approach to long-distance calling. VoIP, or Voice over Internet Protocol, is a technology that takes analog audio signals, such as those produced during phone conversations, and converts them into digital data that can be transmitted via the Internet.
Why is this beneficial? VoIP allows you to transform a regular internet connection into a way to make free phone calls. The key takeaway here is that by using available free VoIP software to make calls, you eliminate the need for a traditional phone service and its associated costs.
VoIP is a groundbreaking technology with the potential to transform the global phone industry. VoIP providers like Vonage have been established for years and are experiencing steady growth. Major telecoms such as AT&T are already introducing VoIP calling options in various U.S. markets, and the FCC is closely examining the impact of VoIP services.
At its core, VoIP is essentially a clever "reinvention of the wheel." In this article, we will delve into the fundamentals of VoIP, its applications, and the possibilities this emerging technology holds, which could one day completely replace the traditional phone system.
What makes VoIP fascinating is that there isn't just one way to place a call. There are three main types or "flavors" of VoIP services that are widely used today:
- ATA -- The most straightforward and commonly used method is with a device known as an ATA (analog telephone adapter). This device allows you to connect a regular phone to your computer or Internet connection for VoIP use. The ATA acts as an analog-to-digital converter, transforming the analog signal from your traditional phone into digital data for Internet transmission. Providers such as Vonage and AT&T CallVantage offer ATAs for free with their services. Simply unbox the ATA, plug in the phone cable that would typically go into the wall socket, and you're ready to start making VoIP calls. Some ATAs come with software that installs on your computer for configuration, but the process is generally simple.
- IP Phones -- These specialized phones resemble standard phones with handsets, cradles, and buttons, but instead of the usual RJ-11 connectors, IP phones have an RJ-45 Ethernet port. They connect directly to your router and come equipped with all the necessary hardware and software to handle VoIP calls. Wi-Fi phones allow users to make VoIP calls from any Wi-Fi hotspot.
- Computer-to-computer -- This is by far the easiest way to use VoIP. No need to worry about long-distance charges. Several companies offer free or inexpensive software for this type of VoIP. All you need is the software, a microphone, speakers, a sound card, and a fast Internet connection, preferably through a cable or DSL modem. Aside from your regular ISP fee, computer-to-computer calls are typically free, regardless of the distance.
If you're interested in exploring VoIP, check out some of the free VoIP software available online. You can download and set it up in just three to five minutes. Get a friend to install the software as well, and you can start experimenting with VoIP to understand how it works.
Next, we will dive into how VoIP is actually used.
How to Use VoIP
VoIP phone users can make calls from any location with a broadband connection.
Photographer: Showface | Agency: DreamstimeChances are, you're already using VoIP whenever you make a long-distance call. Phone companies rely on VoIP to optimize their networks. By routing countless calls through a circuit switch to an IP gateway, they can significantly cut down on the bandwidth used for long-distance communication. Once the call reaches a gateway on the other side, it's decompressed, reassembled, and sent to a local circuit switch.
Though it may take some time, it is inevitable that current circuit-switched networks will be replaced by packet-switching technology (we’ll explore packet switching and circuit switching in more detail later). IP telephony is a smart choice from both an economic and infrastructure perspective. More and more businesses are adopting VoIP systems, and the technology’s popularity will continue to grow as it becomes more accessible for home use. The biggest reasons home users are making the switch are cost and flexibility.
With VoIP, you can place calls from anywhere with broadband access. Since IP phones or ATAs transmit their data over the Internet, they can be managed by the provider wherever there’s a connection. Business travelers can take their phones or ATAs with them on the go and stay connected to their home phone service. Another option is the softphone, which is client software that installs the VoIP service on your desktop or laptop. The Vonage softphone interface appears just like a traditional phone, and as long as you have a headset/microphone, you can make calls from your laptop anywhere with broadband access.
Most VoIP providers offer pay-per-minute plans starting as low as $30 per month, similar to cell phone billing. On the higher end, some companies provide unlimited plans for $79. By eliminating unregulated fees and including a wide range of free features with these plans, VoIP can result in significant savings.
Most VoIP providers include features that traditional phone companies charge extra for when added to your service plan. VoIP services typically offer:
- Caller ID
- Call waiting
- Call transfer
- Redial
- Return call
- Three-way calling
Some carriers offer advanced call-filtering options that use caller ID data to let you choose how calls from specific numbers are handled. You can:
- Forward the call to a designated number
- Send the call straight to voicemail
- Send the caller a busy signal
- Play a "not-in-service" message
- Redirect the caller to a humorous rejection hotline
Many VoIP services also allow you to access voicemail online or send messages to your email, which can be forwarded to your computer or mobile device. Not all VoIP services include all of the above features. Prices and offerings differ, so it’s a good idea to shop around if you're interested.
Now that we’ve covered VoIP in a general sense, let’s dive deeper into the components that make the system work. To truly grasp how VoIP functions and why it’s superior to the traditional phone system, it’s helpful to first understand how the traditional phone system operates.
VoIP: Circuit Switching
Current phone systems operate on a reliable yet somewhat inefficient method called circuit switching to establish calls.
Circuit switching is a fundamental principle that has been the backbone of telephone networks for over a century. When a call is placed, the connection between the two parties is maintained for the entire duration of the conversation. This two-way connection is referred to as a circuit. It forms the core of the Public Switched Telephone Network (PSTN).
Here’s how a typical phone call works:
- You pick up the phone and hear the dial tone, which confirms your connection to your local telephone carrier's office.
- You dial the number you wish to reach.
- Your call is routed through the local carrier's switch to the number you're calling.
- A connection is established between your phone and the recipient's line through a series of switches.
- The recipient’s phone rings, and they pick up the call.
- The connection opens the circuit for the conversation.
- You converse for a while, then hang up the phone.
- Upon hanging up, the circuit closes, freeing up the lines for future calls.
Imagine you're on the phone for 10 minutes. During that time, the circuit stays open the whole duration between both phones. Back in the early days of telephony, up until around 1960, every call required a dedicated wire from one end of the conversation to the other for the entire call. So, if you were in New York calling Los Angeles, the switches would link copper wires across the U.S. for your call. You’d essentially use a 3,000-mile stretch of copper wire for 10 minutes. And you paid a premium for that, essentially owning a long-distance wire for the duration of your call.
Today's traditional phone networks are more efficient and far cheaper. Your voice is digitized, and multiple voices can be sent together over a single fiber optic cable for much of the journey (though your home still uses a copper wire connection). These calls run at a fixed 64 kilobits per second (Kbps) in both directions, for a total of 128 Kbps. Since there are 8 kilobits in a kilobyte (KB), this equals 16 KB per second, or 960 KB per minute. In a 10-minute call, the data adds up to 9,600 KB, or roughly 10 megabytes (see How Bits and Bytes Work for more on these conversions). However, much of the data transmitted during a typical call is essentially wasted.
Next, let's explore packet switching.
VoIP: Packet Switching
A packet-switched phone network is an alternative to circuit switching. Here's how it works: While you’re speaking, the other person is listening, so only half of the connection is actively used at any time. Based on that, we can cut the data in half, reducing it to about 4.7 MB for better efficiency. Additionally, in most conversations, there are pauses where neither party is talking. If we could eliminate these silent moments, the data file would shrink even further. So instead of sending a constant stream of data (including silent gaps), what if we only sent the packets with actual speech?
Data networks don’t rely on circuit switching. If your internet connection kept a constant link to the Web page you’re viewing, it would be far slower. Instead, data networks only send and retrieve information when needed. Rather than traveling along a fixed line, data packets navigate a dynamic network, taking various possible routes. This method is known as packet switching.
Unlike circuit switching, which keeps a constant connection, packet switching only establishes a temporary link—just enough to send a small chunk of data, referred to as a packet, from one system to another. Here's how it works:
- The sending computer breaks the data into small packets, each containing an address that tells the network devices where to send them.
- Each packet carries a payload, which is part of an email, a music file, or whatever data is being transmitted.
- The sending computer then sends the packet to a nearby router, and once it’s sent, it moves on to the next router closer to the recipient. This process repeats until the packet is near the destination.
- Once the recipient computer receives the packets (which may have traveled via different paths), it uses the instructions within the packets to reassemble the original data.
Packet switching is highly efficient. It allows the network to send packets through the fastest and least congested routes, optimizing cost and performance. It also frees up the sending and receiving computers so they can process other data simultaneously.
Next, let’s dive into the benefits of using VoIP.
Benefits of VoIP
VoIP leverages the packet-switching power of the internet to deliver phone services. Compared to traditional circuit switching, VoIP offers many advantages. For instance, packet switching lets multiple calls share the bandwidth that would normally be consumed by just one call in a circuit-switched system. Take that same 10-minute phone call we discussed earlier: under traditional PSTN, it would use 128 Kbps for the full duration. With VoIP, however, that call may only consume minutes at 64 Kbps, freeing up additional bandwidth for more calls. And this calculation doesn’t even include the impact of data compression, which reduces the size of each call even further.
Imagine you and a friend both use a VoIP provider, and each of you has an analog phone connected via the service's ATAs. Let’s walk through a typical call using VoIP over a packet-switched network:
- You lift the receiver, sending a signal to the ATA.
- The ATA responds with a dial tone, confirming an internet connection is established.
- You dial the number you wish to call. The tones are digitized by the ATA and briefly stored.
- The phone number data is sent as a request to the VoIP provider’s call processor, which checks the number’s format.
- The call processor maps the number to an IP address (more on this soon), and the soft switch establishes a link between the two devices. A signal is sent to your friend's ATA to prompt their phone to ring.
- When your friend picks up, a session is established between both computers, meaning each system knows to exchange data packets. The call is routed via the standard Internet infrastructure, similar to how an email or web page is transmitted. Each system communicates using the same protocol, setting up two channels—one for each direction of the conversation.
- During the conversation, both systems send packets when there’s data to transmit. The ATAs translate these packets back into analog audio that you hear. Your ATA also keeps the connection open between your phone and the ATA while forwarding packets between the IP hosts on either end.
- After you finish the conversation, you hang up.
- The circuit between your phone and the ATA is closed.
- The ATA then notifies the soft switch, ending the session.
One of the most significant advantages of packet switching is its compatibility with existing data networks. When telephone systems transition to this technology, they instantly gain the ability to communicate in the same way that computers do.
It will take at least another decade before communications companies can completely transition to VoIP. Like with all new technologies, there are challenges to overcome, and we’ll explore these in the following section.
Challenges of Using VoIP
The current Public Switched Telephone Network (PSTN) is a tried-and-true, resilient system for handling phone calls. Phones just work, and we've come to rely on them. But with computers, email, and related devices, there’s still a bit of unpredictability. A 30-minute email outage might raise an eyebrow, but it doesn’t cause a panic. However, a half-hour of no dial tone is a major crisis for most people. What the PSTN lacks in efficiency, it makes up for in stability. In contrast, the internet is much more complex, and as a result, operates within a wider margin of error. This makes reliability one of the key drawbacks of VoIP.
- VoIP is dependent on electrical power. Traditional phones use phantom power supplied through the phone line from the central office, so even if the power goes out, a standard phone (unless it's cordless) will still work. But VoIP requires an active power source—no power means no phone. A backup power solution is essential for VoIP.
- Another issue is that many other systems in your home may rely on the phone line. Devices like digital video recorders, digital TV services, and home security systems all use the standard phone line to operate. Currently, there's no way to integrate these systems with VoIP, so these industries will need to collaborate to find a solution.
- Emergency 911 calls also pose a challenge with VoIP. Since VoIP uses IP-based phone numbers instead of traditional NANP numbers, it’s impossible to directly associate an IP address with a physical location. If the caller cannot provide their location, it’s difficult to know where to route the emergency call or which EMS team should respond. To address this, geographical information could potentially be integrated into the data packets.
- VoIP, relying on an Internet connection, is prone to issues typical of home broadband services, such as latency, jitter, and packet loss. These factors can degrade call quality, causing distortions or even dropped calls. A reliable internet connection with guaranteed stability is crucial before VoIP can fully replace traditional phone services.
- VoIP can also be vulnerable to security risks, including worms, viruses, and hacking, though this is rare. Developers are working on implementing VoIP encryption to counter these threats.
- Additionally, VoIP systems depend on the performance of individual PCs with varying specs and processing power. For instance, if you’re using a softphone and run a program that drains your system’s processor, it can result in noticeable quality loss. In extreme cases, your system could crash mid-call. VoIP calls are subject to the same limitations as any computer-based application.
One significant obstacle that was solved long ago is the conversion of the analog audio signals received by your phone into digital data packets. How is analog audio transformed into data packets for VoIP? The answer lies in codecs.
VoIP: Codecs
VoIP software handles the processing and routing of calls.A codec, short for coder-decoder, is responsible for converting an audio signal into a compressed digital format for transmission, then reversing the process to play the uncompressed audio. This is the core functionality of VoIP.
Codecs perform the conversion by sampling the audio signal thousands of times each second. For example, a G.711 codec samples the audio at 64,000 times per second, turning each sample into digital data and compressing it for transmission. When these 64,000 samples are reconstructed, the tiny gaps between each sample are so small that, to the human ear, it sounds like a continuous stream of audio. Different codecs use varying sampling rates depending on their design:
- 64,000 samples per second
- 32,000 samples per second
- 8,000 samples per second
The G.729A codec operates with a sampling rate of 8,000 samples per second and is one of the most widely used codecs in VoIP.
Codecs apply sophisticated algorithms to manage sampling, sorting, compressing, and packetizing audio data. A prime example is the CS-ACELP algorithm (Conjugate-Structure Algebraic-Code-Excited Linear Prediction), a commonly used algorithm in VoIP. The CS-ACELP algorithm optimizes the available bandwidth, and its Annex B component establishes a rule that states, 'if no one is speaking, don't transmit any data.' This rule helps maximize efficiency and demonstrates one of the key advantages of packet switching over circuit switching. Annex B is what enables this efficiency in VoIP calls.
While the codec and algorithm handle the data conversion and sorting, the real challenge is knowing where to send the data. In VoIP, this is the job of soft switches.
E.164 refers to the standard used for the North American Numbering Plan (NANP), the system that telephone networks rely on to route calls based on the dialed phone numbers. A phone number essentially functions like an address:
The switches utilize "313" to direct the call to the area code's geographic region. The prefix "555" sends the call to a central office, and the final four digits guide the call to a specific destination. In this system, regardless of your location, dialing "(313) 555" ensures that the call reaches the same central office, which then uses the last four digits, "1212", to pinpoint the correct phone.
The difficulty with VoIP arises because IP-based networks do not process phone numbers like those in the NANP. Instead, they rely on IP addresses, which are formatted as follows:
IP addresses are unique identifiers assigned to devices on a network such as computers, routers, switches, gateways, or phones. However, unlike traditional phone numbers, IP addresses are not fixed. They are allocated by a DHCP server and can change with each new connection. VoIP faces the challenge of converting traditional NANP phone numbers to dynamic IP addresses and locating the current IP address for the requested number. This process is managed by a central call processor that runs a soft switch.
The central call processor is a piece of hardware that operates a specialized database and mapping program known as a soft switch. You can think of the user and the phone or computer as a single entity — man and machine. This combined unit is referred to as the endpoint. The soft switch establishes connections between endpoints.
Soft switches have the following capabilities:
- Identifying the location of the network's endpoint
- Associating a phone number with that endpoint
- Determining the endpoint's current IP address
Next, we will explore more about soft switches and the protocols they use.
VoIP: Soft Switches and Protocols
Customer service call centers, such as this hotline, demand reliable call quality, and many of them depend on VoIP technology for seamless communication.
Tim Boyle/Getty ImagesThe soft switch maintains a user and phone number database. If it can't find the information it's looking for, it forwards the request to other soft switches until it locates one that has the answer. After finding the user, it proceeds to determine the current IP address associated with the user's device, using a similar request chain. The relevant information is then sent back to the softphone or IP phone, enabling data exchange between the two endpoints.
Soft switches collaborate with network devices to enable VoIP functionality. For all these devices to work cohesively, they must communicate using the same language. This communication is a key element that must be refined for VoIP to truly succeed.
Protocols
As we've observed, a VoIP call may involve a mix of analog, soft, or IP phones as user interfaces, ATAs or client software working with a codec to manage digital-to-analog conversion, and soft switches routing the calls. How do we ensure all these diverse pieces of hardware and software communicate effectively to make it all work? The answer lies in protocols.
Several protocols are utilized in VoIP technology. These protocols determine how devices, such as codecs, communicate with each other and with the network. They also set the guidelines for audio codecs. The most prominent protocol is H.323, a standard developed by the International Telecommunication Union (ITU). Originally intended for video conferencing, H.323 is a broad and intricate protocol that outlines specifications for real-time video conferencing, data exchange, and audio applications like VoIP. It is, in fact, a collection of protocols, with each designed for specific tasks.
H.323 Protocol Suite
Video
- H.261
- H.263
Audio
- G.711
- G.722
- G.723.1
- G.728
- G.729
Data
- T.122
- T.124
- T.125
- T.126
- T.127
Transport
- H.225
- H.235
- H.245
- H.450.1
- H.450.2
- H.450.3
- RTP
- X.224.0
H.323 is an extensive suite of protocols and specifications, which makes it suitable for various applications. However, it isn't customized for VoIP.
In response to the limitations of H.323, Session Initiation Protocol (SIP) was developed. SIP is a more efficient protocol, designed specifically for VoIP, and uses existing protocols to simplify the process. Another key protocol is Media Gateway Control Protocol (MGCP), which is focused on endpoint management and functionalities like call waiting. For more information, you can explore Protocols.com: Voice Over IP.
A major challenge for global VoIP adoption is the lack of compatibility between these three protocols. VoIP calls across networks might face issues if they encounter incompatible protocols. As VoIP technology matures, this issue will persist until a universal standard is established.
VoIP offers significant improvements over traditional phone systems in terms of efficiency, cost, and adaptability. While there are challenges to overcome, it’s evident that continuous development will lead to VoIP eventually replacing the current phone system.
The call processor is hardware that operates the soft switch.
VoIP Call Monitoring
VoIP comes with both advantages and drawbacks. The primary benefit is its affordability, while the most significant downside is the quality of calls. This becomes especially problematic for businesses that rely on VoIP, such as call centers (e.g., customer service, technical support, telemarketing), where poor call quality is not acceptable. To resolve call quality issues, many businesses turn to a method known as VoIP call monitoring.
VoIP call monitoring, also referred to as quality monitoring (QM), involves the use of both hardware and software to assess, test, and evaluate the overall quality of calls made via a VoIP network [source: ManageEngine]. This process plays a crucial role in a company's quality of service (QoS) strategy.
To determine the quality of a VoIP call, monitoring systems apply various algorithms to generate a score. The most commonly used score is the mean opinion score (MOS), which is rated on a scale from one to five, although the maximum achievable score on a VoIP network is 4.4 [source: TestYourVoIP.com]. A score of or higher is generally considered a 'good call' [source: ManageEngine].
To calculate the MOS, call monitoring systems analyze several key quality parameters, with the most common ones being:
- Latency – This refers to the delay in communication between the two ends of a VoIP conversation. Latency can be measured as one-way or round-trip delay. Round-trip latency is particularly problematic and leads to the "talk-over effect", where speakers unintentionally interrupt each other because they assume the other person has finished speaking. A round-trip latency of more than 300 milliseconds is typically considered poor [source: TestYourVoIP.com].
- Jitter – Jitter occurs when latency results from packets arriving out of order or too late [source: SearchVoIP.com]. VoIP systems often use a jitter buffer, which groups packets, reorders them, and sends them together to avoid this issue. Callers may notice jitter when it exceeds 50 milliseconds [source: TestYourVoIP.com].
- Packet Loss – A jitter buffer can become overloaded, causing packets to be "dropped" or lost [source: TestYourVoIP.com]. This loss can either be random or result in the dropping of entire sentences during a call (bursty loss) [source: TestYourVoIP.com]. Packet loss is calculated as a percentage of lost packets relative to received ones.
Call monitoring can be categorized into two main types: active and passive. Active (or subjective) monitoring occurs before the VoIP system is deployed. It is usually performed by network specialists and equipment manufacturers who test the system in an isolated environment before it’s made operational. Active monitoring isn't possible once the VoIP system is up and running with employees actively using it [source: VoIP Troubleshooter.com].
Passive call monitoring takes place in real-time during actual user calls. This type of monitoring helps detect network issues, overloads in buffers, and other disruptions that can be addressed by network administrators during non-peak hours [source: VoIP Troubleshooter.com].
Another call monitoring method involves recording VoIP conversations for later review. However, this approach is limited to what is audible during the call and doesn’t provide insight into the network’s performance. This type of monitoring, known as quality assurance, is usually conducted by human reviewers rather than automated systems.
Next, we'll discuss making VoIP calls using cell phones.
VoIP Cell Phones
Dual-mode cell phones feature both a traditional cellular radio and a Wi-Fi (802.11 b/g) radio. The Wi-Fi component allows the phone to connect to the Internet via a wireless router. If you have Wi-Fi at home or find yourself in a coffee shop with wireless access, your phone can make VoIP calls. Here's how it operates:
- When the cell phone detects a nearby wireless network, it automatically connects to it.
- Calls made while connected to the wireless network are routed as VoIP calls. With services like HotSpot@Home, these calls are free of charge.
- If the phone moves out of Wi-Fi range, it switches back to the cellular network, and calls are billed as normal.
- Dual-mode phones allow smooth transitions from Wi-Fi to cellular (and vice versa) during a call, as you move between Wi-Fi zones.
Wi-Fi phones are akin to dual-mode phones, but they differ in that they only feature a Wi-Fi radio and lack a cellular radio. Though they resemble typical cell phones in size and shape, Wi-Fi phones can only make calls when connected to a Wi-Fi network, meaning all calls are VoIP calls.
Wi-Fi phones are especially useful in large companies or offices with extensive Wi-Fi networks. As the market for municipal Wi-Fi grows, these phones may become more significant. Imagine a city-wide high-speed wireless network, offering low-cost or even free VoIP calls no matter where you are. [source: Dr. Dobb's Portal].
In the UK, Hutchinson 3G, also known as 3, has teamed up with the well-known VoIP provider Skype to launch the 3 Skypephone. This phone allows users to make free calls to other Skype users and also supports regular cell phone calls to non-Skype users at standard charges. Here's how it functions:
- To place a Skype call on the 3 Skypephone, you need to be connected to 3's cellular network.
- To start a Skype call, simply locate a Skype contact in your address book and hit the large "Skype" button.
- The call is first routed over 3's GSM network to a fixed Internet connection, which then links to Skype [source: mobileSift].
- From the 3 Skypephone, you can make free VoIP calls to any Skype user, regardless of whether they’re using a Skypephone or another device such as a PC or another Skype-enabled platform.
At present, the 3 Skypephone is not available in the United States.
Use of VoIP in Amateur Radio
Amateur (ham) radio enthusiasts can take advantage of VoIP technology to establish temporary communication stations, such as the one used by the Red Cross in the aftermath of the September 11 attacks.
Tim Sloan/AFP/Getty ImagesImagine amateur radio, or ham radio, as a precursor to the Internet. Using a global network of radio towers, antennas, and transceivers, hobbyists can connect with others worldwide, either through voice communication or even Morse code.
Amateur radio's reach is determined by how far radio waves can travel. Sending a signal across the globe requires precise timing and a bit of luck. For instance, every 11 years, there's a peak in the number of sunspots, boosting the intensity of ionospheric propagation [source: International Solar Terrestrial Physics Program]. This allows ham radio operators to bounce signals off the ionosphere, enabling long-distance communications. During off-peak times, however, reaching far-off locations is much harder.
Amateur radio enthusiasts are now leveraging VoIP technology to connect users globally. Here's how it works: Traditionally, ham radio has relied on FM repeaters, which are large radio towers that act as base stations for accessing the network. By attaching a computer connected to the Internet to these repeaters, users can communicate with the repeater via VoIP.
A number of amateur radio enthusiasts have developed specialized software to connect home radio transceivers to the Internet. Users can hook their transceivers to their PC's sound card, and with the software, they can search for available repeater stations worldwide [source: ARRL]. This breaks the geographical limitation—someone in Indiana can now communicate with a repeater in Mozambique and chat with local ham radio operators in real-time.
There are software programs available that enable communication with other amateur radio users directly from a PC, without needing a physical ham radio [source: ARRL]. While some purists might argue this doesn't count as true amateur radio, others hope this technology will attract a younger audience to the hobby.
For additional insights into VoIP, amateur radio, and related subjects, take a look at the following links.
