Breaking the sound barrier.
A VoIP platform maker uses test equipment to improve its product line.
The transfer of voice traffic over packet networks, and especially voice over IP (VoIP), is rapidly gaining acceptance. Many industry analysts estimate that the over all VoIP market will become a multibillion dollar business within three years.
While many corporations have long been using voice over Frame Relay A high-speed packet switching protocol used in wide area networks (WANs). Providing a granular service of up to DS3 speed (45 Mbps), it has become popular for LAN to LAN connections across remote distances, and services are offered by most major carriers. to save money by utilizing excess Frame Relay capacity, the dominance of IP has shifted most attention from voice over Frame Relay (VoFR) to VoIP. Voice-over-packet transfer can significantly reduce the per-minute cost, resulting in reduced long-distance bills. In fact, many dial-around-calling schemes available rely on VoIP backbones to transfer voice, passing some of the cost savings to the customer. These high-speed backbones take advantage of the convergence of Internet and voice traffic to form a single managed network.
This network convergence also opens the door to novel applications. Interactive shopping (Web pages incorporating a "click to talk" button) is just one example, while streaming audio A one-way audio transmission over a data network. It is widely used on the Web as well as company networks to play audio clips and Internet radio. Computers in home networks stream audio (mostly music) to digital media hubs connected to home theaters. , electronic white-boarding, and CD-quality conference calls in stereo are other interesting applications.
Along with the initial excitement, customers are worried over possible degradation in voice quality when voice is carried over these packet networks. Whether these concerns are based on experience with the early Internet telephony Another term for IP telephony and VoIP. In the late 1990s, some people made a distinction between Internet Telephony and VoIP: Internet telephony referred to voice over the public Internet, while VoIP referred to voice over private IP networks. applications or based on understanding the nature of packet networks, voice quality is a critical parameter in acceptance of VoIP services. Voice quality is not only in the user's interest. It is also in the service provider's interest. Studies done on cellular networks have shown that as voice quality increases, the time users spend on the network also increases. Because users are often billed per use, quality increase translates directly into bottom-line gains for service providers.
HOW DO YOU MEASURE VOICE QUALITY?
With all the factors affecting voice quality--including latency, jitter A flicker or fluctuation in a transmission signal or display image. The term is used in several ways, but it always refers to some offset of time and space from the norm. For example, in a network transmission, jitter would be a bit arriving either ahead or behind a standard clock cycle , packet size, and packet loss--many people ask how to measure voice quality. Standards bodies Following are some of the standards bodies defined in this database. For Windows users of CDE, look up Lessons/Review/Associations. For Web users of CDE's online HTML version, review the Lessons list at the bottom of the definition.
Organization Covers ANSI U.S. like the International Telecommunications Union See ITU.
(body, standard) International Telecommunications Union - (ITU) ITU-T, the telecommunication standardisation sector of ITU, is responsible for making technical recommendations about telephone and data (including fax) communications systems for PTTs and suppliers. are continually addressing this issue and have already delivered two important recommendations: P.800 (MOS (1) (Metal Oxide Semiconductor) See MOSFET.
(2) (Mean Opinion Score) The quality of a digitized voice line. It is a subjective measurement that is derived entirely by people listening to the calls and scoring the results from ) and P.861 (PSQM PSQM Perceptual Speech Quality Measure (voice quality analysis)
PSQM Patient Safety and Quality Monitoring
PSQM Progress in Supersymmetric Quantum Mechanics ). P.800 deals with defining a method to derive a Mean Opinion Score of voice quality. The test involves recording several preselected voice samples over the desired transmission media and then playing them back to a mixed group of men and women under controlled conditions. The scores given by this group are then weighted to give a single MOS score ranging between one (worst) and five (best). An MOS of four is considered "toll-quality" voice.
P.861 Perceptual Speech Quality Measurement tries to automate this process by defining an algorithm through which a computer can derive scores that have a close correlation to the MOS scores. While PSQM is useful, many people have voiced concerns over the suitability of this recommendation to packetized voice The transmission of real time voice in a packet switching network. networks. It seems that PSQM was designed for the circuit-switched network and does not take into account important parameters, such as jitter and frame loss, that are only relevant to VoIP.
As a result of the PSQM limitations, researchers are trying to come up with alternative, objective ways to measure voice quality. One proposal is the perceptual analysis/measurement system (PAMS PAMS
para-amino salicylic acid. ) developed by British Telecom The telephone and communications carrier that provides services in Great Britain and Northern Ireland. It used to be a division of the British Post Office, but was privatized in 1984 under Margaret Thatcher's administration. (BT). Tests conducted by BT have shown good correlation between automated PAMS scoring and manual MOS results.
Sometimes, you have to be your own judge. Testing tools are available that extract individual voice calls and then decompress To restore compressed data back to its original size.
(compression, data) decompress - To reverse the effects of data compression. the captured data using the detected compression method to allow listening to the actual voice recording. By repeating this process at different points along the voice path, it is not only possible to get a sense of quality but also to determine at what point along the network voice-quality degradation occurs.
SIMULATING NETWORK EFFECTS
With so many pitfalls, it is often desirable to simulate the target network before deploying actual equipment--or to test products still in the development stage before they ship to customers. Such simulation allows vendors to gauge the effect of a suboptimal Suboptimal
A solution is called suboptimal if a part of the solution has been optimized without regards to the overall objective. communication link on voice quality and configure critical parameters, such as jitter buffers, before going into the field or sending products to market.
One company using VoIP testing equipment to develop quality product lines and decrease time to market is 3Com Corp. According to according to
1. As stated or indicated by; on the authority of: according to historians.
2. In keeping with: according to instructions.
3. Bert Davenport, a senior product manager at 3Com, the company used a powerful combination of VoIP testing tools from RADCOM RADCOM Radar Communication
RADCOM Radio Communications Company, Inc.
RADCOM Robert and Angela Dianetti Communications
RADCOM Revised Alternative Dataflow Communications
RADCOM Research And Development Communications Equipment to develop CommWorks, a carrier-class IP telephony The two-way transmission of voice over a packet-switched IP network, which is part of the TCP/IP protocol suite. The terms "IP telephony" and "voice over IP" (VoIP) are synonymous. platform that is currently shipping to select customers. RADCOM's AudioPro VoIP Analyzer was combined with RADCOM's PrismLite, a portable and integrated multitechnology protocol analyzer See network analyzer. , to simulate, monitor, and analyze VoIP traffic.
Davenport says that the RADCOM equipment has been extremely helpful in a number of areas, especially in terms of eliminating jitter from the 3Com system, which enables service providers to migrate voice, data, and video traffic onto a single platform to reduce costs and simplify management. "AudioPro has really helped us to design a very robust carrier-class system," he says. "AudioPro allows you to see all the packets and the inner-packet delays, and it does a very nice job of showing you--on a per-stream basis--what the audio stream really looks like on the network."
One feature that Davenport particularly likes is AudioPro's ability to play back actual voice streams. "Sometimes it's nice to hear the audio," he says. "The AudioPro can translate what you have into an audible .wav file The native digital audio format in Windows. Using the .WAV file extension, 8- or 16-bit samples can be taken at rates of 11,025 Hz, 22,050 Hz and 44,100 Hz. The highest quality (16-bit at 44,100 Hz) is the sampling rate of an audio CD and uses 88KB of storage per second. . In combination with the PrismLite, it's great to see the protocol. If we had a bad protocol or a bad protocol message, that's really nice to see"--and hear.
The PrismLite's portability, combined with AudioPro's simple GUI (Graphical User Interface) A graphics-based user interface that incorporates movable windows, icons and a mouse. The ability to resize application windows and change style and size of fonts are the significant advantages of a GUI vs. a character-based interface. , has extra advantages.
"We've used it not only for testing but also for customer demonstrations," says Davenport. We've taken it into the field to show customers the quality of our system. It's especially nice when you have a lot of data streams, because if you just have one callup, you can do this stuff with a regular protocol analyzer. But if you bring up one span, or five spans, or 10 spans of calls, you've got a lot of traffic on your network and you can't separate out each stream easily. The AudioPro separates out each stream as an independent entity."
The future of IP telephony is a converged network, says Davenport, where a single network is going to be carrying voice, data, and video traffic. "It's not going to happen overnight," he says. "But, eventually we're going to get to the point where the entire infrastructure, or a significant portion of it, is IP-based."
And that's great news for companies like 3Com that are building next-generation VoIP solutions to provide integrated products and services to users. It's also great news for behind-the-scenes testmakers like RADCOM, who are building not only valuable products but also a loyal customer base.
"At the time we made our [AudioPro] purchase, there was nobody else in the market," says Davenport. "There may be a few other players in the market now, but we haven't really looked at them. We've been too happy with the RADCOM AudioPro."
Circle 268 for more information from RADCOM Equipment, Inc.
Yuval Boger, vice president, voice test and management, RADCOM Equipment, Inc., contributed to this article.