VoIP Testing and Management Methods
The team at Codima understands the necessity for VoIP testing and analysis. As organizations assess and then start to deploy VoIP in earnest on their networks they face two major inter-related challenges:
- An economic challenge: How to test VoIP and manage the roll-out of this young and sometimes problematic technology in an affordable way?
- A technical challenge: What is the most practical way to monitor VoIP and identify the causes of low voice quality for a large number of IP phones and calls.
VoIP Testing Methods for Voice Quality Measurement
Subjective Testing
The "human" method of measuring voice quality is a specialized and costly process which averages the results of human evaluations of call quality to produce a mean opinion score (MOS). A MOS score ranges from 1 for an unacceptable call to 5 for an excellent call. A satisfactory range for VoIP is between 3.5 and 4.2.
Active Testing
P.861/PSQM (Perceptual Speech Quality Measure) and the newer P.862 analyze the distortion on test voice signals transmitted through a VoIP network to produce an estimated MOS score. It is an intrusive method that requires calls to be set up between specific agents on the network. Its focus is on measuring the quality between two defined end-points rather than individual IP phones or calls.
Passive Monitoring
Passive or non-intrusive monitoring examines a stream of voice traffic and produces a transmission quality metric that can be used to estimate a MOS score. This method monitors all IP phones and calls in a network without any additional network overhead. The method calculates an ‘R’ factor on a scale of 1-100 based on a number of variables such as equipment impairment, jitter, delay and other elements. Scores of 70 to 94 are considered good with anything less than 50 being unsatisfactory.
Cost of deployment and effectiveness in problem resolution are the two most important criteria for most organisations when choosing a VoIP testing and management system. Codima’s autoVoIP™ products addresses these concerns directly. It also introduces the concept of One-Click Management™, which both integrates multiple functions, and makes them extremely easy to use.
Key autoVoIP™ Architectural Elements
Incorporated into autoVoIP™ are key architectural and design elements specifically for VoIP testing. These include:
- Automatic monitoring of RTCP, which is a non-intrusive and easy to deploy mechanism to monitor voice calls. It also has the advantage that it reports on end user experiences rather than between test probes placed on the network.
- Passive monitoring of RTP and performance statistics regarding network operations such as bandwidth utilisation, response times and protocol utilisation.
- Integration of SNMP, which is the best way to look at the internal performance records of network infrastructure devices such as switches and routers.
- Integration with a topology discovery function, which is a major time-saver when trying to identify the source of quality degradation issues. It determines which network devices in the transmission path are relevant for problem analysis purposes.
- An automated correlation function, which sifts through the multiple sets of data to achieve a deterministic result for root-cause analysis and problem resolution.
- Zero to minimal configuration from the user.
- Access to all functions by either clicking on a browser-like tab to sort or filter results, or by clicking on an icon to view more detailed reports.
In sum, autoVoIP™ supports both a cost-effective way of gathering the necessary data to monitor call-quality levels and integrates the means to identify the likely causes of call quality degradation. It automatically correlates its sources of data for effective problem determination and with One-Click Management™, truly redefines ease-of-use for a multifunction product.
The autoVoIP™ Traffic Simulator addresses the growing need for traffic simulation to ensure successful implementation of VoIP networks. Our VoIP testing software can be used pre- and post-deployment and measures Quality of Service (QoS) at different points on the network by simulating synthetic phone traffic.
- Uses real RTP frames, sent at standard frame rates over UDP with RTP ports. That ensures the QoS engineering in the network will treat the stream the same as normal VoIP Traffic - in contrast to an ICMP Pinger.
- With the Traffic Blaster tool, sends out competitive and non-competitive traffic to see what the QoS would be for a single user when different levels of VoIP and non VoIP traffic are present.
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