Showing posts with label codec. Show all posts
Showing posts with label codec. Show all posts

Monday, January 30, 2012

Call Center Contact Center Phone Service

Call centers and contact centers are major users of telephone services, especially long distance and toll free number services. What’s the best option for your call center to ensure that you have the right features and the best pricing? Well, there are several good technical options available right now. Let’s take a closer look at them to see what they can offer.

Check out options and pricing for call center and contact center phone service ...Traditional telephony has been based on basic business phone service, also called POTS or Plain Old Telephone Service. Old is right. Analog phone lines have been used for over 100 years and are still going strong. They’re highly reliable, deliver clear voice quality and don’t suffer from digital artifacts like latency.

Analog lines were originally intended to be one line per telephone. Key systems allow multiple lines on each phone so you can select which one to answer manually. All the but the smallest call centers have more lines and more agents than a key system can handle. That means using a PBX or Private Branch Exchange to automatically assign lines and direct incoming calls.

What PBX offers is automation. If you only need a half dozen or so outside lines, analog business lines still make economic sense. You connect them to your PBX using FXO (Foreign Exchange Office) ports on interface cards that plug into your PBX system.

When you need more than a half dozen or so outside lines, digital trunk service starts to make sense. Why? It’s because there is a cost advantage to bundling multiple phone lines into one trunk line. You may already have multiple analog lines coming in via a multi-pair binder cable. This is different. A T1 line uses two twisted copper pair to carry 23 or 24 separate phone lines digitally. Yes, those are the same copper pair that would otherwise be used for analog phone service.

The way this works is that the phone conversations are digitized at one end using an analog to digital converter and converted back to analog at the other end using a digital to analog converter. Each call is assigned its own digital timeslot or channel. They are kept separate so that there is no cross-talk between calls. T1 is a synchronous system that has strict timing at both ends of the line. Latency is almost non-existant. However, any unused channels are transmitted empty. All channels are constantly being exchanged between your PBX and your telephone service provider.

Why the choice between 23 and 24 phone conversations on a T1 line? All 24 channels can be used for telephone calls. This is called a T1 telephone line or T1 trunk. The other option is to use 23 channels for conversations and use one channel for switching, signaling and data, such as ANI and Caller ID. That option is called ISDN PRI or T1 PRI. PRI trunks are very popular with call centers because they offer the Caller ID information and faster switching times than normal T1 phone lines. Your PBX system may have multiple PRI ports so that you can have 23, 46, 69 or more outside lines for your call center.

SIP Trunking is an alternative to ISDN PRI. It is based on VoIP technology and uses packets rather than channels. SIP Trunks can be interfaced to older technology PBX systems and the newer IP PBX phone systems. There may be cost advantages for SIP Trunking, especially if using a lower bandwidth CODEC (Coder/Decoder) than the industry standard G.711. Some newer CODECs offer high call quality while transporting more simultaneous calls on the same line bandwidth.

SIP Trunking also opens the opportunity for cloud communications or Hosted PBX. With a hosted system, both the outside lines and the PBX switching system move to the cloud. You have only phones and a VoIP gateway in-house. The advantage to a cloud hosted solution is that you avoid heavy capital investment, pay for service by the agent seat per month, and don’t have to worry about maintenance or upgrades. The service provider takes care of all this for you.

Are you starting up a new call center or contact center, facing an upgrade cycle or expanding operations? If so, this is the perfect time to check business phone service options and prices for call centers and contact centers. You could gain performance advantages at lower prices now.

Click to check pricing and features or get support from a Telarus product specialist.




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Thursday, August 25, 2011

How HD Audio Will Transform VoIP

Those of us old enough to remember when phones were analog and switches were mechanical have a soft spot for the clarity of the old school system. There was a distinctness and intimacy to those old time conversations that has long been lost. Well, perhaps lost only for some decades. There’s a new move afoot to give VoIP phone conversations a starting fidelity that will make today’s handsets and cellphones seem like toy walkie talkies by comparison.

High definition audio takes VoIP way beyond anything that has come before...The technology is HD or high definition audio. It’s called HD Voice by Polycom, the telephone manufacturer producing IP telephones with this capability built-in. The idea is to reverse the steady decline in voice intelligibility that we’ve experienced with the digital transformation and create a new standard that is superior to anything that has come before.

There’s a parallel here. Something similar happened in the radio industry. If you’ve ever owned one of those big console radios from the 1930’s or even the table models from the 50’s, you know that AM radio can sound amazingly lifelike. The advent of transistor radios shrunk the size of the loudspeakers, and thus the range of sound frequencies that can be heard. “Tinny” was the word to describe the loss of low end or bass notes. Car radios added their own restrictions by rolling off the high frequencies to reduce noise and static. It took a new technology, FM radio, to expand the frequency range and eliminate the background noise before the term high fidelity could be applied.

Old-timey phones had very few parts. The carbon microphone and electromagnetic earphone were powered by a battery at the phone company and connected to a similar phone at the other end. There were no electronics in the circuit. The telephone company switch was really a mechanical switch that simply connected phones together. The resulting sound had a richness that made local conversations much closer to someone sitting next to you than you hear today.

I remember when I first noticed the difference. It was when the company I worked for upgraded their old phone system with a new digital PBX and desksets. The first calls seemed slightly more muffled, less sparkling clear on the new phones. The effect was subtle, but noticeable. There was no noise or distortion apparent, but something was missing.

What was missing, I later figured out, was some of the audio range in the conversation. It wasn’t digital that degraded phone audio specifically. It started with long distance calling. The analog carrier multiplexing used to load multiple conversations onto copper wire and microwave trunk lines needed filters to keep one conversation from interfering with another. After much research, a frequency range of 300 to 3400 Hz was selected as standard. Frequencies above 3400 Hz add to naturalness, but don’t limit understandability of speech very much. Get rid of them and you can transmit more telephone channels on a trunk. You also get rid of annoying hiss, which is a high frequency phenomena. Frequencies below 300 Hz also aren’t necessary for intelligibility, but certainly give voices and other sounds their distinctiveness. By getting rid of the low frequencies, problems with power line hum getting into phone lines are greatly reduced or eliminated.

This standard was maintained when the industry changed from analog to digital. It kept everything compatible and consistent. Digital technology was a boon to long distance calls, especially overseas. It totally eliminated the noise and crosstalk that were common in the analog era. That advantage went away with cell phones, which can be noisy or even drop out.

Early implementations of VoIP only made matters worse. Packet transmission added distortion from lost packets and variable transmission time or jitter. Latency, a time delay through the network, was almost unknown with analog and TDM telephony. Now it’s not uncommon on poorly implemented systems to have a clipping effect so that you can’t both talk at the same time.

What HD audio intends to do is to lift telephone voice quality to a new standard above and beyond what we’ve ever had. This is accomplished by using a wideband CODEC (Coder/Decoder) to handle the conversion from analog microphone and earpiece to digital transmission. Bandwidth is doubled to 7 kHz with the new standard G.722 Codec and can be extended up to 20 Khz, the same quality as CDs and digital FM radio.

HD Voice is here now. You can get it with Polycom business phones and a service provider like RealLinx that can provide the hosted PBX and carefully managed bandwidth to maintain high quality phone audio. Of course, this only works within an organization that is set up for HD Voice or among users that all have the G.722 Codec available. When one party is using traditional telephone technology, everything defaults to the universal G.711 telco standard and its limited frequency range.

Even so, higher fidelity within your organization can improve conversation intelligibility, including nuances in the voice that are often lost when talking on the phone rather than in person. This can be valuable when budgets and time restrictions reduce face to face meetings in favor of audio and video teleconferencing.

Are you ready to replace an aging phone system or order equipment for a new enterprise or new location? If so, why limit yourself to yesterday’s technology when you can have something that will provide a competitive edge at lower overall cost than you have now? Don’t make a move until you get pricing and features on HD Voice Hosted PBX VoIP telephone systems.

Click to check pricing and features or get support from a Telarus product specialist.




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Monday, July 13, 2009

What is ISDN and Why Do I Want It?

You may have heard the term ISDN related to telephone services. So what is it and what can it do for you?

ISDN stands for Integrated Services Digital Network. It's a fancy term for digital telephone service. But not just any digital phone service. This one has very specific standards.

The story behind ISDN goes back to the days when the standard for Internet access was a dial-up modem using an ordinary analog telephone line. ISDN was envisioned as a way for users to have a combination of separate telephone service and Internet service on a single ISDN telephone line with three channels. Two of these are called "B" or bearer channels and are 64 Kbps each. The third is the "D" or delta channel that is used for switching and signaling. One B channel can be used for a telephone handset. The other can be used for a modem. Or both channels can be combined for 128 Kbps Internet service. This arrangement is called ISDN BRI. The BRI means Basic Rate Interface.

The telephone companies expected ISDN BRI to be a hit with consumers as an upgrade to their analog phone service. But by the time it was ready for mass deployment, Internet standards had moved on and everybody really wanted DSL or Cable broadband. After all, 128 Kbps may be impressive when compared to a dial-up modem, but it pales in comparison with always-on broadband at 512 Kbps or higher. Today, even mobile users want bandwidth above 1 Mbps, so ISDN BRI is of little interest to most users.

The exception is in the field of broadcasting and professional audio. ISDN BRI lines can be used to transmit high quality audio through the telephone network by using a CODEC (Coder / Decoder) to convert the audio to and from the ISDN format at each end of the call. Guests around the country can join a panel discussion and sound like they are right in the studio instead of hundreds or thousands of miles away. Voice over artists can contribute to a production from their home studios.

Where ISDN flourishes today is in its second standard, called PRI or Primary Rate Interface. ISDN PRI consists of 24 channels configured as 23 bearer channels plus 1 delta channel. The 23 bearer channels can each support 1 telephone call each. That's 23 digital phone lines, plus the delta channel used for signaling and Caller ID data.

ISDN PRI is the telephone service of choice for medium and large company telephone systems. Most PBX and IP PBX business phone systems offer built-in support for at least one ISDN PRI line. Other systems can be configured for one or more ISDN PRI trunks with a plug-in circuit card.

In addition to office PBX telephone, ISDN PRI is preferred by company contact centers and independent call centers. There is a significant cost reduction realized by using a single ISDN PRI instead of a dozen or more individual analog phone lines. Each channel on the PRI can be configured to be local and/or long distance, incoming and/or outgoing calls. If more than 23 lines are needed, more can be added to the system to double, triple or quadruple the number of active phone lines.

ISDN PRI is delivered on a T1 line in the United States. For this reason, it is sometimes known as T1 PRI or T1 telephone service. In other countries, an E1 line might be used to provide PRI service.

Now that you know what ISDN is good for, can this service help your business or organization to provide high quality voice communications for perhaps less than you are spending now? Find out by checking ISDN PRI service pricing now.

Click to check pricing and features or get support from a Telarus product specialist.




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Tuesday, April 29, 2008

Why Video is Moving to H.264

It is well known that video applications are sopping up Internet bandwidth almost as fast as it can be deployed. Digital video is the hot new medium of communications. It's taking over television broadcasting, becoming a larger part of Web content, and being looked at as combination cost savings, productivity improvement and "green" enabler for corporations. Video has many things going for it and one big drawback. It's a bandwidth hog. In fact, it's a bandwidth monster.

A video coding standard called H.264 offers the opportunity to get better quality video at lower bandwidths. Yes, it's a compression standard. Hardly anybody can afford to have uncompressed high quality video running around their networks at anywhere from a couple hundred Mbps to over a Gbps. That's the realm of video producers and national broadcast networks. For other businesses and consumers, video compression is necessary to fit within the bandwidth of reasonably priced wireless and wireline transport systems. The real trick in compression is to reduce the bandwidth requirement while preserving the quality of the original video stream.

The popularity of H.264 is growing because it maintains perceived video quality while cutting bandwidth requirements by as much as half. It does this by using a sophisticated algorithm within its CODEC (Coder/Decoder) hardware and software. The coders and decoders aren't some simple circuitry. There's some very sophisticated processing going on in there to determine which bits can be dropped without anyone noticing and which much be preserved to maintain scene quality.

You'll be hearing a lot more about H.264 and see it popping up in the technical specifications of all sorts of equipment and applications. Cisco's TelePresence service, a high definition version of traditional video conferencing, uses H.264. So do HTDV broadcasters in Europe. Blu-Ray high definition DVDs are H.264 based. Apple has incorporated it as a standard in its QuickTime video player.

What H.264 has going for it is worldwide standardization and the ability to tailor its processing to accommodate a wide range of video requirements from cell phone displays, embedded players in Web pages all the way up to high end video conferences and HD videos. You'll also find it referred to as MPEG-4 Part 10 or MPEG-4 AVC (Advanced Video Coding). That's different than the earlier MPEG-4 Part 2 standard and certainly different than the MPEG-2 which is widely used for standard TV satellite transmission. Satellite TV services such as Dish Network and DirecTV use MPEG-2 for standard TV channels but MPEG-4 AVC for their newer HDTV channels.

If video is becoming an important part of your business tools or part of your creative content, you may already be feeling the pinch of limited bandwidth. Newer compression technology, such as H.264, may help ease the congestion. So will transport bandwidth upgrades that are less expensive now than ever before. Find out how Ethernet, bonded T1, and SONET fiber optic services can help you keep up with business bandwidth demands.

Click to check pricing and features or get support from a Telarus product specialist.




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