Q: What is Digital Voice and how does it work?

A: Digital Voice is VoIP wich stands for “Voice over Internet Protocol.” It means voice communication over the Internet rather than over the traditional publicly switched telephone network (PSTN). In the same way that your computer turns your keyboard typing into an e-mail and transmits it via the Internet Protocol (IP) standard, it can also turn voice data into a form that is transmitted via IP and reassembled at the receiver’s computer or, increasingly, a specially equipped landline or mobile phone. By 2010, Gartner also predicts that IP-telephony products will represent 90 percent of new phone system sales. Traditional PBX systems are clearly in decline as customers opt for more open, feature rich, hybrid IP PBX solutions.

Q: What is IP PBX?

A: An IP PBX takes the place of the PBX you may already have for your company’s PSTN calls. Like its PSTN cousin, an IP PBX (for private branch exchange) is an electronic switchboard that receives, routes, holds, forwards to voice mail, or otherwise manipulates calls that arrive over the Internet, rather than via the PSTN. It may be fully automatic or have a human receptionist who routes incoming calls from a main IP phone number to internal IP numbers or extensions. Where a PSTN PBX can connect many incoming and internal phone lines through a set of mechanical or electronic switches, an IP PBX will be mechanically simpler, typically either software that resides on a server or a small, independent server that connects with your existing data network.

Q: What are the advantages of IP PBX?

A: An IP PBX provides more efficient switching and a more professional “look” than if everyone in a business has their own separate IP connection and account. It allows phone calls to be forwarded within the company, and for internal voice-mail and conferencing capabilities that might otherwise have to be outsourced. An IP PBX is also much more flexible than a PSTN PBX, allowing an essentially infinite number of extensions and voice-mail boxes, plus desktop management via a Web browser rather than at a set of PSTN switches. They can also enable the recording of incoming and outgoing conversations (subject to legal considerations). IP PBXs, both as software and as physical devices, are relatively inexpensive and can be had for as little as $1,000.

Q: What is hosted IP PBX?

A: Hosted IP PBX is a service provided off-site by a third-party company; with it, the IP PBX, associated equipment, and the responsibility for maintaining and upgrading the PBX lie with the hosted IP PBX provider.

Q: How can our company start using an IP PBX?

Q: Why is the legacy PBX market in decline?

A: The emergence of open standards and protocols has created a new model in telephony where customers are increasingly

in control. More recently, open source technology, such as Asterisk, has driven real technology innovation and parity between vendor and buyer when it comes to economic relationships. In the legacy market, PBX phone system vendors thrived in a world where they could sell proprietary systems, drive customers to single vendor environments, create customer lock-in via proprietary call control and endpoints, and force high margin phone and system deals on to customers.

Q: What are the things that affect my Digital Voice Quality of Service (QoS)?

A: A variety of problems can affect Digital Voice Quality of Service, including dropped packets (too much data arrives at the receiving server too quickly), packet delay (data takes the long way around the Internet), jitter (different packets reach the receiver at different times) and related out-of-order delivery, and other mishandling of the data packets themselves. Each problem causes delays, which in time-sensitive settings like voice leads to lowered voice quality or even the dropping of whole calls.

Q: What is latency and why is it important on a Digital Voice Network?

A: Latency is the length of time it takes for your words to be received by a listener at the other end of a phone connection, typically expressed in milliseconds. According to a white paper from Brooktrout Technology, latency starts to affect phone conversations when it exceeds 150 milliseconds each way, and is unacceptable when it exceeds 450 milliseconds (nearly half a second). The company recommends engineering a Digital Voice system so that latency is always below 200 milliseconds and suggests some ways to accomplish this task here.

Q: What are some ways to measure the performance of a Digital Voice network?

A: Latency is the total of a lot of small delays that occur during the coding, transmission and decoding of a voice conversation. The time each step takes can be measured, and then improved through better network equipment and better configuration of the network. Performance factors within a business’s control include the presence of absence of a firewall for Digital Voice traffic;

delays in the digital signal processor (DSP) that codes and decodes voice traffic; how big a chunk of data the DSP takes on at once; how big a buffer you build in against delays elsewhere in the Digital Voice system; and how efficiently the decoded signal is routed within your data network to an IP-enabled phone or PC.

Performance factors outside a business’s control include the length of the path packets travel between computers across the Internet, and the speed at which signals travel through physical lines or satellite links (typically a high fraction of the speed of light)..

Q: What is Asterisk?

A: In the same way that Linux is an open-source computer operating system, Asterisk is an open-source IP PBX. Like Linux, Asterisk has an enthusiastic developer community committed to improving it. It is available on a large number of platforms, including Linux, Mac OS X, OpenBSD, FreeBSD and Sun Solaris. Asterisk is free, can interoperate between Digital Voice and traditional PSTN systems, and supports the protocols these two systems use, including H.323 (a PSTN protocol), Session Initiation Protocol (SIP), Media Gateway Control Protocol (MGCP), and Skinny Client Control Protocol (SCCP).

Q: What is SIP?

A: Session Initiation Protocol (SIP) is a set of standards that helps create, change and end IP sessions between one or more users, such as in the case of Digital Voice calls or audio conferences. It is notably lightweight and transport-independent, and was designed by the IP-community programmers rather than by the telecom industry.

E-mail: Digital Voice Information