The VoIP PABX or IP PABX

An IP or VoIP PABX handles voice signals under Internet protocol, bringing benefits for computer telephony integration (CTI – Computer Telephony Integration). An IP PABX can exist as physical hardware, or can carry out its functions virtually, performing the call-routing activities of the traditional PABX as a software system. The virtual version is also called a “Soft PABX”.

VoIP, or Voice over Internet Protocol, is a method for taking analog audio signals, like the kind you hear when you talk on the phone, and turning them into digital data that can be transmitted over the Internet.

How is this useful? VoIP can turn a standard Internet connection into a way to place free phone calls. The practical upshot of this is that by using some of the free VoIP software that is available to make Internet phone calls, you’re bypassing the phone company (and its charges) entirely.

VoIP is a revolutionary technology that has the potential to completely rework the world’s phone systems.

Above all else, VoIP is basically a clever “reinvention of the wheel” which will more than likely one day replace the traditional phone system entirely.

The interesting thing about VoIP is that there is not just one way to place a call. There are three different types of VoIP service in common use today:

• ATA The simplest and most common way is through the use of a device called an ATA (analog telephone adaptor). The ATA allows you to connect a standard phone to your computer or your Internet connection for use with VoIP. The ATA is an analog-to-digital converter that takes the analog signal from your traditional phone and converts it into digital data for transmission over the Internet.
• IP Phones These specialized phones look just Like normal phones with a handset, cradle and buttons. IP phones connect directly to your router/switch and have all the hardware and software necessary right onboard to handle the IP call.
• Computer-to-Computer This is certainly the easiest way to use VoIP. You don’t even have to pay for long-distance calls. There are several companies offering free or very low-cost software that you can use for this type of VoIP. All you need is the software, a microphone, speakers, a sound card and an Internet connection, preferably a fast one like you would get through a cable or DSL modem. Except for your normal monthly ISP fee, there is usually no charge for computer-to-computer calls, no matter the distance.

Phone companies use VoIP to streamline their networks. By routing thousands of phone calls through a circuit switch and into an IP gateway, they can seriously reduce the bandwidth they’re using for the long haul. Once the call is received by a gateway on the other side of the call, it’s decompressed, reassembled and routed to a local circuit switch.

Although it will take some time, you can be sure that eventually all of the current circuit-switched networks will be replaced with packet/switching technology. IP telephony just makes sense, in terms of both economics and infrastructure requirements. More and more businesses are installing VoIP systems, and the technology will continue to grow in popularity as it makes its way into our homes and business. Perhaps the biggest draws to VoIP for the users that are making the switch are price and flexibility.

With VoIP, you can make a call from anywhere you have broadband connectivity. Since the IP phones or ATA broadcast their info over the Internet, they can be administered by the provider anywhere there’s a connection. So business travelers can take their phones or ATA with them on trips and always have access to their home phone. Another alternative is the softphone that is a client software that loads the VoIP service onto your computer. As long as you have a headset/microphone, you can place calls from your laptop anywhere in the broadband-connected world.

VoIP Packet Switching

 

A packet-switched phone network is the alternative to circuit switching. It works like this: While you’re talking, the other party is listening, which means that only half of the connection is in use at any given time. Based on that, we can surmise that we could cut the file in half. Plus, a significant amount of the time in most conversations is dead air – for seconds at a time, neither party is talking. If we could remove these silent intervals, the file would be even smaller. Then, instead of sending a continuous stream of bytes (both silent and noisy), what if we sent just the packets of noisy bytes when you created them?

Data networks do not use circuit switching. Internet connection would be a lot slower if it maintained a constant connection to the Web page when one was viewing at any given time. Instead, data networks simply send and retrieve data as needed. And, instead of routing the data over a dedicated line, the data packets flow through a chaotic network along thousands of possible paths. This is called packet switching.

While circuit switching keeps the connection open and constant, packet switching opens a brief connection – just long enough to send a small chunk of data, called a packet, from one system to another.

Packet switching is very efficient. It lets the network route the packets along the least congested and cheapest lines. It also frees up the two computers communicating with each other so that they can accept information from other computers, as well.

VoIP technology uses the Internet’s packet-switching capabilities to provide phone service and has several advantages over circuit switching. Probably one of the most compelling advantages of packet switching is that data networks already understand the technology.

By migrating to this technology, telephone networks immediately gain the ability to communicate the way computers do.

It will still be at least some years before telecommunications companies can make the full switch over to VoIP – as with all emerging technologies, there are certain hurdles that have to be overcome and investments to be planned, respected and profitably.

Disadvantages of Using VoIP

 

The current Public Switched Telephone Network (PSTN) is a robust and fairly bulletproof system for delivering phone calls. Phones just work, and we’ve all come to depend on that. On the other hand, computers, e-mail and other related devices are still kind of flaky.

Let’s face it – few people really panic when their e-mail goes down for 30 minutes. It’s expected from time to time. On the other hand, a half hour of no dial tone can easily send people into a panic.

So, what the PSTN may lack in efficiency it more than makes up for in reliability.

The network that makes up the Internet is far more complex and therefore functions within a far greater margin of error, what it’s one of the major flaws in VoIP: reliability.

First of all, VoIP is dependant on electric power – current phone runs on phantom power that is provided over the line from the central office – even if power goes out, common phone still works. With VoIP, no power means no phone – a stable power source must be “created” for VoIP.

Another consideration is that many other systems may be integrated into the phone line: fax, security systems and others, all using a standard phone line to do their job and there’s currently no “reliable” way to integrate these products with VoIP.

Emergency calls also become a challenge with VoIP – as stated before, VoIP uses IP addressed phone numbers, not real phone numbers, so there’s no way to associate a geographic location with an IP address; so, if the caller can’t tell the Emergency operator where he is located, then there’s no way to know which call center to route the emergency call to and which Emergency Medical Service should respond – to fix this, perhaps geographical information could somehow be integrated into the VoIP packets in the future.

Because VoIP uses an Internet connection, it’s susceptible to all the hiccups normally associated with broadband services; all these factors affect call quality:

• Latency This is the time delay between two ends of a VoIP phone conversation. It can be measured either one-way or round trip. Round-trip latency contributes to the “talk-over effect” experienced during bad VoIP calls, where people end up talking over each other because they think the other person has stopped speaking. A call with a round-trip latency of over 300 millisecond is considered a bad call;
• Jitter Is latency caused by packets arriving late or in the wrong order. Most VoIP networks try to get rid of jitter with something called a “jitter buffer” that collects packets in small groups, puts them in the right order and delivers them to the end user all at once. VoIP callers will notice a jitter if latency is of 50 millisecond or greater;
• Packet loss Part of the problem with a jitter buffer is that sometimes it gets overloaded and late-arriving packets get “dropped” or lost. Sometimes the packets will get lost sporadically throughout a conversation (random loss) and sometimes whole sentences will get dropped (bursty loss). Packet loss is measured as a percentage of lost packets to received packets.

So, phone conversations can become distorted, garbled or lost because of transmission errors – some kind of stability in Internet data transfer needs to be guaranteed before VoIP could truly replace traditional phones.

Another issue associated with VoIP is having a phone system dependant on computers of varying specifications and power – a call can be affected by processor drain: let’s say you are chatting away on your softphone and you decide to open a program that saps the processor – quality loss will become immediately evident and even, in a worst case scenario, your system could crash in the middle of an important call – in VoIP, all phone calls are subject to the limitations of normal computer issues.

VoIP Codecs

 

One of the hurdles that were overcome some time ago was the conversion of the analog audio signal your phone receives into packets of data. How it is that analog audio is turned into packets for VoIP transmission? The answer is: codecs.

A codec, which stands for coder-decoder, converts an audio signal into compressed digital form for transmission and then back into an uncompressed audio signal for replay. It’s the essence of VoIP.

Codecs accomplish the conversion by sampling the audio signal several thousand times per second – for instance, a G.711 codec samples the audio at 64,000 times a second. It converts each tiny sample into digitized data and compresses it for transmission. When the 64,000 samples are reassembled, the pieces of audio missing between each sample are so small that to the human ear, it sounds like a continuous audio signal.

There are different sampling rates in VoIP depending on the codec being used:
G.711 – 64,000 times per second
G.728 – 32,000 times per second
G.729A – 8,000 times per second

A G.729A codec has a sampling rate of 8,000 times per second and is the most commonly used codec in VoIP.

Codecs use advanced algorithms to help sample, sort, compress and packetize audio data.

The CS-ACELP (conjugate-structure algebraic-code-excited linear prediction) algorithm is one of the most prevalent algorithms in VoIP, it organizes and streamlines the available bandwidth and it’s Annex B creates the transmission rule, which basically states: “if no one is talking, don’t send any data” – the efficiency created by this rule is one of the greatest ways in which packet switching is superior to circuit switching.

The codec works with the algorithm to convert and sort everything out, but it’s not any good without knowing where to send the data. In VoIP, that task is handled by soft switches.

The challenge with VoIP is that IP-based networks don’t read real phone numbers, they look for IP addresses, which look like this: 192.158.10.7

IP addresses correspond to a particular device on the network like a computer, a router, a switch, a gateway or a telephone; however, IP addresses are not always static, most of the times they’re assigned by a DHCP server on the network and change with each new connection.

VoIP’s challenge is translating phone numbers to IP addresses and then finding out the current IP address of the requested number – this mapping process is handled by a call processor running a soft switch.

The call processor is hardware that runs a specialized database/mapping program called a soft switch – think on the user and his phone or computer as one package (man + machine), that package is called the endpoint, the soft switch connects endpoints.

Soft switches know:

• Where the network’s endpoint is;
• What phone number is associated with that endpoint;
• The endpoint’s current IP address.

The soft switch contains a database of users and phone numbers and, if it doesn’t have the information it needs, it hands off the request downstream to other soft switches until it finds one that can answer the request; once it finds the user, it locates the current IP address of the device associated with that user in a similar series of requests, it sends back all the relevant information to the softphone or IP phone, allowing the exchange of data between the two endpoints.

Soft switches work in tandem with network devices to make VoIP possible and, for all these devices to work together, they must communicate in the same way – this communication is one of the most important aspects that will have to be refined for VoIP to take off.

Protocols

 

As we’ve seen, on each end of a VoIP call we can have any combination of an analog, proprietary, soft or IP phone as acting as a user interface, ATA or client software working with a codec to handle the digital-to-analog conversion, and soft switches mapping the calls.

How to get all of these completely different pieces of hardware and software to communicate efficiently to pull all of this off? The answer is: protocols.

There are several protocols currently used for VoIP. These protocols define ways in which devices like codecs connect to each other and to the network using VoIP and they also include specifications for audio codecs.

The most widely used protocol is H.323, a standard created by the International Telecommunication Union (ITU), but it wasn’t specifically tailored to VoIP.

An alternative to H.323 emerged with the development of Session Initiation Protocol (SIP).

SIP is a more streamlined protocol, developed specifically for VoIP applications. Smaller and more efficient than H.323, SIP takes advantage of existing protocols to handle certain parts of the process.

Media Gateway Control Protocol (MGCP) is a third commonly used VoIP protocol that focuses on endpoint control that is geared toward features like call waiting.

One of the challenges facing the worldwide use of VoIP is that these three protocols are not always compatible; VoIP calls going between several networks may run into a snag if they hit conflicting protocols.

Since VoIP is a relatively new technology, this compatibility issue will continue to be a problem until a governing body creates a standard universal protocol for VoIP.

VoIP is a vast improvement over the current phone system in efficiency, cost and flexibility and, like any emerging technology, VoIP has some challenges to overcome, but it’s clear that this technology will be keep refining until it, we believe, will replace the current phone system.