Dynamic VoIP Routing for Audio Quality Management
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Solution Overview
Problem
VoIP audio quality is compromised due to packet loss and delays, with existing systems relying on subjective human feedback for quality improvement, leading to inefficient and costly processes that cannot retroactively address unsatisfactory call quality.
Innovation Solution
A dynamic VoIP audio quality management mechanism that monitors audio quality in real-time during calls, redirecting VoIP traffic and re-negotiating parameters such as RTP socket type, payload size, and encoding algorithms to improve call quality without requiring human intervention.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Productivity
If VoIP packets are transmitted over IP networks with fixed maximum capacity, then network bandwidth is efficiently utilized, but audio quality deteriorates due to packet loss and congestion
Solution Approach 1:
The patent implements dynamic routing that adjusts packet transmission paths in real-time based on network conditions. The system monitors audio quality metrics and automatically redirects packets through alternative routes when congestion or packet loss is detected, transforming the static routing into a dynamic adaptive system that maintains audio quality while utilizing network bandwidth efficiently
Solution Approach 2:
The system changes transmission parameters dynamically by negotiating and adjusting VoIP parameters such as RTP socket type, payload size, and encoding algorithms based on real-time audio quality assessment. This allows the system to adapt packet transmission characteristics to current network conditions, improving reliability without sacrificing bandwidth utilization
2Reliability
If human feedback is used to improve call quality, then audio quality can be optimized, but the process becomes inefficient and cannot address quality issues in real-time
Solution Approach 1:
The system implements self-service quality management by automatically monitoring audio quality metrics and initiating corrective actions without human intervention. The automated system assesses call quality in real-time and dynamically adjusts routing and parameters to maintain satisfactory audio quality, eliminating the need for post-call human feedback and enabling immediate quality improvements
Solution Approach 2:
The patent implements real-time automated feedback mechanisms that continuously monitor audio quality metrics during calls. This feedback loop enables the system to detect quality degradation immediately and trigger dynamic routing adjustments or parameter changes without waiting for human feedback, thus eliminating the time delay inherent in manual quality assessment
3Reliability
If dynamic routing adjustments are made during ongoing calls, then audio quality can be improved immediately, but system complexity increases
Solution Approach 1:
The patent introduces intermediary components such as media servers and border controllers that facilitate dynamic routing adjustments. These intermediaries manage the complexity of real-time routing changes and parameter negotiations, enabling quality improvements during ongoing calls while abstracting the complexity from the end-user system
Data Source
AI summary
Embodiments described herein provide a dynamic voice over Internet Protocol (VoIP) audio quality management mechanism in real time, e.g., when a VoIP call is ongoing. Specifically, when a VoIP call has unsatisfactory audio quality, e.g., due to packet loss, jitter, etc., the dynamic VoIP audio quality management mechanism may redirect the VoIP traffic from the previous endpoint that initiates the VoIP session to a different endpoint within the same carrier. Upon the endpoint redirection, a new call leg is established, allowing re-negotiation or re-configuration of VoIP parameters. The re-negotiated or re-configured VoIP parameters may then be used to conduct the remainder of the VoIP call to improve the audio quality.