VOIP Call Quality Calculation with Mid-Call Codec Changes
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Solution Overview
Problem
Existing methods for monitoring call quality in Voice Over IP (VOIP) communications fail to accurately account for mid-call codec changes, which affect the quality of voice communications by altering packet delivery and bit rates, leading to unclear impacts on quality metrics like Mean Opinion Score (MOS).
Innovation Solution
A system and method that determine VOIP call quality by using time-weighted averages of Ie-eff values before and after codec or bit rate changes, calculating Ieff values for burst and gap transitions, and applying these to estimate the MOS, ensuring accurate quality assessment even with mid-call modifications.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If conventional data network metrics (packet loss, delay, jitter) are used to measure VOIP quality, then network performance is captured, but voice quality from listener perspective is not accurately reflected
Solution Approach 1:
The patent transforms network-level parameters (packet loss, delay, jitter) into voice-quality-related parameters by applying codec-specific impairment models. This converts raw network metrics into meaningful voice quality estimates that reflect listener experience, addressing the loss of voice quality information while maintaining network performance measurement capabilities.
Solution Approach 2:
The patent introduces an intermediary layer of codec-specific impairment models that mediate between network metrics and voice quality assessment. These models act as translators that convert network-level measurements into voice-quality estimates, bridging the gap between network performance and user experience without requiring direct listener feedback.
2Adaptability or versatility
If mid-call codec changes occur to adapt to network conditions, then network adaptability is improved, but quality metric calculation becomes unclear and inaccurate
Solution Approach 1:
The patent segments the VOIP call into distinct temporal segments based on codec changes and network conditions. Each segment is analyzed separately with its own quality metrics, allowing accurate measurement despite mid-call adaptations. This segmentation enables the system to maintain measurement precision while accommodating network adaptability through continuous re-evaluation of quality parameters.
Solution Approach 2:
The patent implements dynamic quality assessment that adapts to changing network conditions in real-time. The system continuously updates quality metrics based on current codec settings and network state, transforming static measurement approaches into dynamic processes that maintain accuracy during mid-call codec changes while preserving network adaptability.
3Reliability
If codec changes are implemented to optimize packet delivery, then packet loss and delay are reduced, but the impact on voice quality metrics becomes complex and difficult to measure
Solution Approach 1:
The patent applies codec-specific impairment models that translate packet-level metrics (loss, delay, jitter) into voice-quality-specific parameters. This parameter transformation simplifies the measurement of quality impact by converting complex packet delivery dynamics into manageable voice quality estimates, reducing measurement difficulty while maintaining reliability assessment.
Solution Approach 2:
The patent implements feedback mechanisms that continuously monitor network conditions and adjust quality assessments accordingly. By incorporating real-time feedback from network status and codec performance, the system simplifies the measurement of quality metrics through adaptive algorithms that account for the relationship between packet delivery reliability and perceived voice quality.
Data Source
AI summary
The embodiments described herein relate to obtaining one or more measurements from a VOIP communication that is indicative of call quality, changing one or more settings of the VOIP communication, such as the type of codec or the ascribed bit rate of a VBR codec, and determining call quality of the VOIP communication after setting changes by utilizing at least one of the one or more call quality measurements.