The introduction of VoIP (Voice-over-IP and other media-over-IP) or IP Telephony can have major performance effects on existing networks and applications, and QoS implications for routed networks. To be useful VoIP must also deliver good voice quality, usually expressed as a MOS score (mean opinion score) or measured to PAMS, PESQ or PSQM+ standards.
To avoid implementation problems and surprises, pre-deployment testing and performance benchmarking is highly recommended if not essential. Post-implementation, VoIP is another application requiring protocol analysis and troubleshooting, not only for its own performance but also for the effects of VoIP data on overall network performance. Phoenix Datacom supplies a range of scaleable products appropriate to both pre-and post-deployment testing.
Pre-deployment Testing Pre-deployment testing reduces the risk by allowing you to run realistic and"worst-case" testing in a controlled way. Traffic generation, network traffic analysis and IP network simulation are vital tools that help reduce the risk.
Traffic Generation
Agilent N2X systems stress-test equipment, including switches, routers and server load balancers, networks and services by producing multi-port, multi-interface traffic streams with total control over all traffic parameters such as packet size and variation, errors and losses and data rates. Load modules cater for all major LAN, WAN, ATM and SAN interfaces from 10/100 ethernet up to 40Gigabit.
Agilent Network Tester systems provide the most powerful and flexible solution for testing the performance of Layer 4-7 devices and services. NetworkTester creates a realistic mix of application traffic through its unique ability to simultaneously emulate real voice, video, data and multiple DOS, spam and virus attacks on a single test port. Coupled with the ability to simulate proprietary protocols like P2P, gaming and IM, NetworkTester provides the most realistic application performance system in the market.
Navtel Interwatch systems provide an extremely powerful test system for next generation networks, such as Voice over IP and Fixed Mobile Converged (IMS) networks, and related equipment such as Session Border Controllers, Media Gateway Controllers and Softswitches.
The strength of Navtel's Interwatch lies in its ability to combine VoIP signalling emulation (SIP, Megaco, PacketCable, NCS, IPSEC, DQoS) with RTP data generation and voice quality testing - over very large numbers of user agents and IP addresses.
A single test chassis can maintain a call rate of nearly 2000 complete calls per second with 128,000 simultaneous RTP streams - allowing testing with realistic call set-up and call hold times, reflecting the "real world" rather than theoretical tests.
Arca Harmony systems provide bulk call generation for capacity, functional and inter-operability testing of telephony systems and components

IP Network Simulation
Perceived voice quality on VoIP systems is severely affected by packet loss, packet delay and jitter (variable packet arrival times) - all factors inherent to any packet-based system. The solution is to "tune" networks, giving higher priority to VoIP traffic, but this can adversely affect the performance of other applications! Changing the parameters of a live network can be a time-consuming and disruptive process. The alternative is to use a network simulator - a device that allows you to accurately model the target network and then see the effect of changing key parameters such as routing, link speeds, priorities etc
Packetstorm Network Emulator systems provide a "complete IP network in a box" which can be used stand-alone or in-circuit with a real network to allow consistent and worst-case testing of networked applications. The Packetstorm system simulates a complete packet-based network with multiple elements in the traffic path (LAN, Router, modem, WAN link etc), each of which can be set with specific parameters for impairments, modifiers and traffic conditioning. Advanced features include:- emulation of Routing & Bridging, Diff Serv with traffic conditioning, ToS emulation, IP monitoring, packet counters and timers, Tcl scripting language, network capture & replay, packet modifiers, multiple LAN, WAN and ATM network interfaces, network queues, and remote control.
Applications Performance Analysis
Without proper tuning, the introduction of VoIP will adversely affect the performance of other applications running on the same network - so benchmarking applications performance before you start is essential! OPNET ACE Live is ideal for that task and allows you to benchmark performance on the existing network by individual user, logical groups of users across the organisation, or by application. Critically, ACE Live shows which elements of user-perceived applications performance are caused by the network, the server and the application itself.

Protocol Analysis
It is most important to measure traffic levels and benchmark the performance of the existing network before introducing a new VoIP application. This provides an objective basis for any performance tuning or problem troubleshooting that may be necessary after implementation. Agilent Network Analyzer provides expert analysis of a huge range of VoIP and non-VoIP protocols and is an excellent tool for pre-deployment benchmarking. Because Network Analyzer is available in software only, stand-alone hardware or distributed versions with a wide range of LAN, WAN and ATM interfaces, it is scaleable to the size of the project, from departmental segments to core backbones.

Post-deployment Testing
These questions arise after VoIP implementation:-
- Is the VoIP system delivering adequate perceived voice quality? (And if not, why not?)
- Is the VoIP application affecting the performance of other applications? (And if so, why?)
Answering these questions effectively will save time, money and arguments! Voice Quality Testing and Troubleshooting
Voice is, by definition, analogue in nature and voice quality is highly subjective and the same network may deliver different perceived quality for different voice types. When it comes to measuring voice quality, there are two basic approaches - analogue and digital.
An analogue voice quality tester works by generating a real voice call across a network and comparing the received call with the reference call. This gives the most objective independent test of voice quality, (whether IP based or conventional telephony), and the results are measured to established standards such as PAMS, PESQ or PSQM+
A digital voice quality tester works by analysing the packet data stream containing the VoIP data and measuring packet loss, packet delay and packet jitter (variation in inter-packet arrival time); this information is then used to calculate the likely voice quality, expressed as predictive Mean Opinion Score (MOS).
Depending on the scale and nature of the VoIP project, one or other or both methods may be appropriate. The Agilent VQT provides comprehensive testing across analogue, IP-based, and hybrid telephony systems. By including testing for voice clarity, delay, echo, silence suppression, DTMF, and signal loss, all on one box, the VQT provides a total solution for testing voice quality on next generation networks.
The Agilent VQT supports analog, T1, E1, and Ethernet interfaces to enable testing at different access points for network segmentation and fault isolation. Using different signaling types, including FXO, E&M, CAS, ISDN, SIP, and H.323, the VQT enables testing across any telephony network. VQT is also available in a software-only version for testing via a standard ethernet NIC card.
A good network analyser is essential for troubleshooting problems, which may show themselves in any part of the network. Agilent Network Analyzer provides expert analysis of a huge range of VoIP and non-VoIP protocols and is an excellent tool for pre-deployment benchmarking. Because Network Analyzer is available in software only, stand-alone hardware or distributed versions with a wide range of LAN, WAN and ATM interfaces, it is scaleable to the size of the project, from departmental segments to core backbones. The Telephony Network Analyzer provides expert analysis of VoIP related data and provides a predictive MOS score for voice quality on IP networks. Used as a software option with any of the Network Analyzer family, this effectively allows voice quality to be sampled at any point of a LAN/WAN/ATM network. This is vital in pinpointing the cause of problems.
The NeTracker product from Sunrise Telecom ia an extremely versatile unit that
provides accurate signaling and traffic simulation, protocol analysis, and Quality of Service (QoS) measurements
over both legacy and NGN networks
Rapid assessment and troubleshooting of operational VoIP links is provided by the VoIP options on the FrameScope Pro handheld tester for 10/100/1000 ethernet links and the SunSet MTT handheld modular comms tester.

Applications Performance QOS Monitoring
VoIP as an application introduces new protocols on to the network and additional data traffic that may affect its own and other applications' performance. Traffic prioritisation to achieve good VoIP performance can also adversely affect the performance of other applications. Monitoring of the actual Quality of Service (QoS) and Quality of Experience (QoE) as perceived by users is essential - to provide a performance benchmark when troubleshooting problems -and to avoid a lot of finger-pointing between different suppliers of different parts of an overall IT solution!
ACE Live for VoIP is designed to monitor VoIP performance right across an organisation and report by individual user, logical groups of users across the organisation, or by location. Critically, ACE Live shows which elements of user-perceived applications performance are caused by the network, the server and the application itself. Insight for VoIP works alongside the other features of ACE Live(see Applications Performance Analysis above) so that it is also to monitor and assess the interaction between VoIP and other applications.
TAMS from Sunrise Telecom provides probe-based continuous monitoring of both signaling and data across networks and can report voice qualityfor both legacy PSTN and VOIP.

To find out more about VoIP related products and services, call Phoenix Datacom on 01296 397711 , send an email or use the Request More Info form.
For a full list of products and applications, click here. |