Web Conference Audio Quality via Probe Packet Latency

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Solution Overview

Problem

Web-conferencing applications experience poor audio quality over low-speed internet connections due to bandwidth contention between VoIP audio and data share streams, leading to issues like silence, gaps, distortion, and excessive latency, which existing methods such as RSVP, differentiated services, and TCP congestion control fail to adequately address in unmanaged networks.

Innovation Solution

The method involves transmitting probe packets before and after data packets to measure latency differences, allowing the web-conferencing application to adjust transmission parameters, specifically reducing datashare component transmission to mitigate audio quality degradation by controlling data flow without requiring managed network infrastructure.

Engineering Contradictions & Design Principles

VSEngineering Contradiction Analysis

1Productivity

If data share streams are transmitted over the network connection, then data transmission capability is improved, but audio quality deteriorates due to bandwidth contention

Engineering Contradiction:

Improvedata transmission capabilityVSAvoidaudio quality

Core Design Contradiction:

ProductivityVSReliability

Solution Approach 1:

The system dynamically adjusts the transmission rate of data share packets based on real-time network conditions and audio packet delivery quality. When audio quality degrades, the data transmission rate is reduced to prioritize audio traffic, and when network conditions improve, data transmission is increased to utilize available bandwidth. This dynamic adjustment resolves the contradiction by making data transmission adaptive rather than static.

Inventive Principle:

Principle #15Dynamics

Solution Approach 2:

The system implements a feedback mechanism where the delivery quality of audio packets is monitored and used to control the transmission rate of data share packets. The receiver sends feedback about audio packet loss or delay, and the transmitter adjusts data share transmission accordingly. This closed-loop control ensures that data transmission does not compromise audio quality while maximizing overall network utilization.

Inventive Principle:

Principle #23Feedback

2Use of energy by moving object

If multiple data streams share the same network connection, then network resource utilization is improved, but serialization delay increases causing audio quality degradation

Engineering Contradiction:

Improvenetwork resource utilizationVSAvoidserialization delay

Core Design Contradiction:

Use of energy by moving objectVSLoss of time

Solution Approach 1:

The system dynamically controls the transmission rate of data share streams based on real-time measurement of serialization delay impact on audio packets. When serialization delay exceeds thresholds that would degrade audio quality, the data transmission rate is reduced or paused temporarily. This dynamic rate control allows multiple streams to share the connection efficiently while preventing excessive serialization delay.

Inventive Principle:

Principle #15Dynamics

Solution Approach 2:

The system takes preliminary action by monitoring network conditions and proactively adjusting data share transmission before audio quality degradation occurs. By detecting early signs of serialization delay or bandwidth contention, the system reduces data transmission rates in advance to prevent audio packet loss or excessive delay, rather than reacting after degradation has occurred.

Inventive Principle:

Principle #9Preliminary anti-action

3Ease of operation

If bandwidth management mechanisms like RSVP or TCP congestion control are used, then network traffic control is improved, but effectiveness deteriorates in unmanaged networks

Engineering Contradiction:

Improvenetwork traffic controlVSAvoidcontrol effectiveness

Core Design Contradiction:

Ease of operationVSReliability

Solution Approach 1:

The system implements self-service bandwidth management where the web conferencing application itself monitors its own network traffic patterns and audio quality, and autonomously adjusts its data share transmission rate without requiring external network management infrastructure. The application measures the impact of its own data packets on audio delivery and self-regulates to resolve contention, making the system effective in unmanaged networks where protocols like RSVP cannot enforce control.

Inventive Principle:

Principle #25Self-service

Data Source

PatentUS9667686B2System and method for improving audio quality during web conferences over low-speed network connections

Publication Date: 2017.05.30 CISCO TECHNOLOGY INC

AI summary

A method that includes: (1) transmitting, at a first transmit time point, a first probe packet over a network connection to a conferencing server immediately before transmitting a data packet, the first probe packet arriving at the conferencing server at a first receive time point; (2) transmitting, at a second transmit time point, a second probe packet over the network connection to the conferencing server immediately after transmitting the data packet, the second probe packet arriving at the conferencing server at a second receive time point, the first and second probe packets being smaller than the data packet; (3) receiving information encoding a first difference between the first and second transmit time points and a second difference between the first and second receive time points; and (4) based on the first and second differences, modifying a transmission parameter associated with data packets to be transmitted thereafter to the conferencing server.