Voice Quality Assessment Using Network Impairment Data

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Solution Overview

Problem

Current methods for assessing voice quality in packet networks, such as VOIP or SIP, are either intrusive, impractical for large networks, or do not provide true end-to-end measurements, particularly when it comes to network latency and packet loss.

Innovation Solution

A non-intrusive, parameter-based method that calculates the Mean Opinion Score (MOS) using information from Cisco gateways, specifically considering codec, network delay, and packet loss, to dynamically reroute voice calls when quality falls below a predetermined threshold.

Engineering Contradictions & Design Principles

VSEngineering Contradiction Analysis

1Measurement precision

If intrusive method is used to measure voice quality by injecting reference signal, then true end-to-end quality assessment is achieved, but the method becomes intrusive and requires additional receivers

Engineering Contradiction:
Improvevoice quality assessment accuracyVSAvoidnetwork infrastructure complexity
Core Design Contradiction:
Measurement precisionVSDevice complexity

Solution Approach 1:

The patent introduces a quality assessment function that acts as an intermediary by utilizing existing network elements (gateways, routers) to collect impairment data without requiring additional reference signal injection or dedicated receivers. The function computes voice quality metrics by processing available network performance data, thereby achieving end-to-end quality assessment without the complexity of intrusive methods.

Inventive Principle:
Principle #24Intermediary (Mediator)

2Measurement precision

If signal-based non-intrusive method is used to assess voice quality by listening to phone calls, then true end-to-end measurements are obtained, but the method does not account for network delay and becomes impractical for large networks

Engineering Contradiction:
Improveend-to-end voice quality measurementVSAvoidnetwork scalability
Core Design Contradiction:
Measurement precisionVSDevice complexity

Solution Approach 1:

The patent segments the voice quality assessment into two complementary parts: (1) network impairment data collection from existing infrastructure elements at different network points, and (2) quality metric computation by the quality assessment function. This segmentation allows the system to account for network delay and scale to large networks while still providing end-to-end measurements, as each segment processes local data that contributes to the overall quality picture.

Inventive Principle:
Principle #1Segmentation

3Device complexity

If parameter-based non-intrusive method is used to assess voice quality by collecting impairment information, then the method is non-intrusive and scalable, but it does not give true end-to-end results

Engineering Contradiction:
Improvenetwork scalabilityVSAvoidend-to-end voice quality accuracy
Core Design Contradiction:
Device complexityVSMeasurement precision

Solution Approach 1:

The patent merges multiple impairment parameters (packet loss, delay, jitter, codec information) collected from different network points into a unified voice quality metric through the quality assessment function. By combining these segmented impairment measurements with the E-model formula, the system achieves true end-to-end voice quality assessment while maintaining the scalability and non-intrusive nature of parameter-based methods.

Inventive Principle:
Principle #5Merging (Combining)

Data Source

PatentUS7756052B2Determining quality of voice calls over a packet network
Publication Date: 2010.07.13 ITXC IP HLDG R L
  • US7756052B2 patent drawing
  • US7756052B2 patent drawing
  • US7756052B2 patent drawing

AI summary

The present invention provides a method for managing a voice call over a packet switched network, in which the voice call is rerouted when voice quality is below a predetermined level. In particular, the voice quality is determined by calculating a parameter representing the voice quality based on information regarding codec, network delay and packet loss. Preferably, R-factor and MOS is calculated based on the information obtained from Cisco CDRs during the call session.