August 2008 | Volume 11/ Number 8
Design for Superior Voice Quality
By: Richard “Zippy” Grigonis
Achieving the best possible voice quality is everyone’s goal. Network Equipment Manufacturers (NEMs) fret over it from early-stage unit testing in R&D labs through verification in trials and pilot projects. Application developers ponder what algorithms will yield the best Quality of Service (QoS). Service providers wanting to deploy a new packet-based service find themselves immersed in end-to-end testing, from proof-of-concept, service verification, interoperability, and scenario testing. Businesses and government organizations now perform network assessments prior to deploying VoIP systems to ferret out any packet traffic performance problems that could impact voice quality.
Ditech Networks (News - Alert) is a leading provider of voice quality solutions to conference providers, wireline and mobile carriers, cable operators, managed service providers, interconnect and peering providers to solve churn and customer satisfaction problems that result from substandard voice quality.
Ken Croley, Ditech’s Senior Director of Marketing, says, “We sell hardware for both TDM/PSTN and IP-type networks. Our hardware is loaded with software consisting of algorithms deployed on DSP [Digital Signal Processor (News - Alert)] chips and we run every single phone call through the box and then the algorithms act on that phone call to scrub it of what are called voice quality impairments. We do the same thing in both TDM and IP networks. Our devices can be deployed in a wireless network and an IP network for VoIP. That’s what we do.
Ditech’s Packet Voice Processor [PVP] optimizes and measures voice quality in real-time, on every phone call, in both directions for VoIP and 3G networks. PVP can non-intrusively detect and eliminate problems emanating from the three things that degrade voice quality: network-induced impairments, impairments from callers’ devices, and impairments from the caller’s environment.
“In the IMS [IP Multimedia Subsystem (News - Alert)] architecture, our PVP is considered a Media Resource Function [MRF],” says Croley. “And then we load it with software called Voice Quality Assurance [VQA], which has six algorithms that do everything from noise reduction to echo cancellation to packet restoration. That’s essentially what the box does in an IP network.”
“We find that VoIP either sounds great or it’s terrible,” muses Croley. “Strangely, there seems to be little middle ground in terms of quality. We’ve all experienced calls that are great and those that are not-so-good, sometimes in the same call. The early adopter market for our products in an IP environment is the conferencing industry, where you may have an IP bridge, and people call into that bridge using different mobile devices and codecs from different networks. The complication in that mix is that many voice quality impairments can be introduced that wreck conference calls. InterCall (News - Alert), a subsidiary of the West Corporation and the world’s largest conferencing service provider, uses our VQA solution to eliminate noise, echo and to deal with packet loss problems on large and small VoIP conference calls throughout its IP network. That’s important because when you’re in the middle of a conference call and you’re perhaps negotiating a contract, you shouldn’t suddenly miss a word or experience an annoying echo that will disrupt the call. It’s not a ‘nice thing to have’, it’s a hard requirement.”
“We’re also approached by cable companies to deal with acoustic echo,” says Croley. “Acoustic echo is caused by mobile phones and Bluetooth headsets, or it can be caused simply by your voice bouncing off of the windshield of your car and coming back into the microphone of your cellphone. That’s a common problem across a lot of networks. It becomes pronounced when you begin to interact with different kinds of mobile devices. So, we receive calls to correct and solve these problems too.”
“There are several other issues that can impact voice quality in an IP environment,” says Croley. “These can range from the codecs deployed to the location where you place the phone call, to the device you’re using, to packet delay caused by different parts of the network, to bandwidth constraints based on either access or what’s going on in the core network. So there are many issues that have the potential to impact voice quality, and we can deal with them.”
Whether designing an application, service or network, testing has become a fundamental part of development, deployment and monitoring processes. Empirix (News - Alert), for example, has over the years devised a huge assortment of testing and monitoring products and services that have enabled both businesses and service providers to successfully migrate to new voice technologies, spanning computer telephony, TDM/PSTN, VoIP, NGN and IMS-based networks. Empirix’ testing and monitoring solutions can be found in everything from R&D labs to deployments where they monitor the end-user experience.
Normally, ensuring quality in VoIP, NGN, and IMS devices and networks involves many types and stages of testing, each with its own unique test requirements and personnel. But instead of maintaining many pieces of test equipment throughout the product development and service deployment lifecycle, you can simply buy Empirix’ Hammer G5, combining feature testing, scalable load testing, voice quality, signaling and media under one common GUI. The Hammer G5’s comprehensive test environment can be used throughout the test lifecycle for application developers, network equipment manufacturers, and service providers.
The Empirix Hammer G5 supports and can run multiple protocols simultaneously using the same tests and scripts, including: SIP, MGCP, NCS, H.323, QSIG, and Cisco’s (News - Alert) Skinny. It also supports many different narrowband and wideband codecs for wireline and wireless multimedia, as well as various real media types such as tones, DTMF, voice, fax, and video. Vendors and service providers often rely on the Hammer G5 for all forms of end-to-end testing (proof of concept, service verification, interoperability, etc.). The Hammer G5 can simulate thousands of distinct and concurrent subscriber interactions with the network using multiple protocols.
Then there’s the carrier-class Hammer XMS, designed for VoIP monitoring of next-gen networks. The Hammer XMS was adopted, for example, by PowerNet Global (News - Alert) Communications (PNG), a provider offering various integrated voice, data, and Internet solutions nationwide to residential and commercial customers. PNG sought a VoIP monitoring solution that would allow it to take a proactive approach to VoIP troubleshooting, fix problems more quickly, improve the call quality of VoIP traffic, and reduce costs. It turned to Hammer XMS.
The Hammer XMS integrates diagnostic, analytic and monitoring capabilities for signaling and media quality. It can monitor all calls, 24x7, for VoIP, SS7 and ISDN signaling and VoIP (RTP) media. It’s also highly scalable; using distributed high-performance probes and centralized data management. It supports Gigabit wire-rate processing for IP packet filtering, and its open architecture is loaded with full set of SNMP MIBs defined to interface with existing network management systems. It can correlate multiple call legs across both VoIP and TDM protocols in an IMS-enabled network, and it can drill down from high level call analysis to individual call protocol message decodes.
Empirix’ Hammer XMS is complemented by Hammer XMS “Active”, which is designed to add active monitoring, so that VoIP service providers can now have even greater insight into customer QoS. The Hammer XMS can even deal with examining the more advanced features offered by service providers in their service bundles, such as three-way calling and fax services. Hammer XMS Active has two components: Active probes to emulate IP and TDM voice and fax endpoints; and an Operations Server for applications, such as probe management, scheduling tests, diagnostics, reporting and system administration.
GL Communications (News - Alert) also provides testing products and professional telecom engineering and IT consulting services to the worlds of cellular, wireless, microwave, fiber-optic, T1 and satellite communications. GL’s analysis and emulation test products fall into three broad areas: TDM, VoIP wireless.
Recently, GL Communications released their Improved Voice Quality Testing Solutions (VQT) Application. GL’s VQT combined with their VQuad™ software enables sophisticated voice quality testing of VoIP equipment, yielding results that include PESQ/PAMS/PSQM+ and Mean Opinion Scores (MOS).
The VQuad with VoIP option provides the ability to perform manual or automated tests on a VoIP network. The VQuad provides direct connection to the VoIP network with up to eight instances connected simultaneously (i.e., up to eight independent tests using different interfaces can be executed simultaneously). Users may configure automatic call control (via SIP protocol) along with automatic generation/reception of voice files. Path confirmation using digits or frequency tones is also available in an automated or manual operation.
The GL VQT employs three algorithms to perform the voice comparisons: the Perceptual Evaluation of Speech Quality (PESQ LQ/LQO) per Rec. P.862/P862.1, the Perceptual Analysis / Measurement System (PAMS) per Rec. P.800, and the Perceptual Speech Quality Measurement (PSQM) per Rec. P.861. PESQ provides an objective measurement of subjective listening tests on telephony systems. PAMS predicts overall subjective listening quality (a human’s perception of quality) without requiring expensive actual subjective testing. PSQM predicts subjective quality of speech codecs without requiring subjective testing. The GL VQT performs PESQ, PAMS, and PSQM (+) simultaneously, using two voice files (Reference File and Degraded File) and provides the algorithm results along with analytical results in both a graphical and tabular format.
Thus, VQuad with VQT software provides a single-box solution to automatically establish calls (PSTN, VoIP, Wireless, T1/E1), send/record voice over the established call, and perform voice quality analysis on the recorded voice files. The measurement results are reported both locally and remotely through GL’s NetViewer and/or web-based VQT WebViewer applications.
Paving the Way for Voice Quality
These days, for any organization about to implement an IP communications system, an infrastructure pre-assessment has become commonplace, even mandatory. In some cases, IP phone system vendors themselves can assist you. For example, ShoreTel (News - Alert) is a well-known provider of IP unified communications solutions based on a scalable, distributed architecture administered via a friendly interface. A ShoreTel Network Assessment is required prior to installing a new ShoreTel IP telephony system across multiple sites. Their Network Assessment is quite comprehensive and helps you plan, design and implement an optimal IP telephony solution. The assessment can be administered by a ShoreTel solutions partner or by a ShoreTel engineer.
The ShoreTel Network Assessment monitors both LAN and WAN links over several days. Aside from revealing conditions that degrade IP telephony performance, such a long assessment can identify the source of intermittent problems. WAN monitoring is included since wide area connections are often the source of latency, jitter, and packet loss issues.
Of course, once you have an optimal next-gen network through careful testing and analysis, there’s no reason why you can’t boost the quality of your resulting communications even more by installing systems based on wideband, high-fidelity voice codecs. VoiceAge (News - Alert) is a company that has some considerable expertise in this area, offering hi-fi telephony capabilities powered by their VoiceAge G.722.2/VMR-WB wideband speech codecs that expand the transmitted speech spectrum to a range of 50-7,000 Hz so as to provide a richer, super-realistic sound quality. The VoiceAge familiy of codecs can be found deployed in wireline, wireless and WiFi (News - Alert) networks for VoIP, conferencing, and multimedia applications. Their codecs have undergone rigorous comparative testing across multiple languages and operating conditions. Moreover, unlike other solutions, the VoiceAge G.722.2/VMR-WB wideband speech codecs provide transcoder-free interoperability across wireline and wireless networks worldwide – preserving top quality even at very low bit rates and in adverse network conditions.
Thus, whether you’re developing a VoIP product or service or are simply planning to purchase one for your company, a vast array of techniques and procedures are available for achieving superior voice quality. IT
The following companies were mentioned in this article:
Ditech Networks (www.ditechnetworks.com)
GL Communications (www.gl.com)
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