VoIP Bandwidth Management via Receiver Feedback
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Solution Overview
Problem
Voice-over-Internet-Protocol (VoIP) systems face challenges in optimizing audio quality due to varying network conditions, as existing methods struggle to accurately determine the optimal bitrate for audio transmission, leading to potential congestion and increased latency in interactive sessions.
Innovation Solution
A computerized method that involves the receiver calculating one-way delay and trend using double-exponential smoothing to determine congestion, estimating available bandwidth, and sending this estimate to the sender to adjust the maximum allowed send rate, thereby optimizing audio quality by adjusting bitrate based on network conditions.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If the bitrate is increased to improve audio quality, then audio quality is improved, but network congestion and latency increase
Solution Approach 1:
The patent implements dynamic bitrate adjustment by continuously monitoring network conditions (one-way delay, packet loss, available bandwidth) and adapting the audio bitrate in real-time. The receiver calculates current bandwidth availability and sends feedback to the sender, which then adjusts the transmission bitrate dynamically. This resolves the contradiction by making the bitrate flexible rather than fixed, allowing high quality when network conditions permit and lower quality when congestion occurs.
Solution Approach 2:
The patent employs a feedback mechanism where the receiver measures network conditions (including one-way delay and available bandwidth) and communicates this information back to the sender. The sender uses this feedback to adjust the bitrate accordingly. This closed-loop control system resolves the contradiction by continuously adapting to network conditions, preventing congestion-induced latency while maintaining optimal audio quality.
2Reliability
If the bitrate is decreased to reduce congestion, then network congestion is reduced, but audio quality deteriorates
Solution Approach 1:
The system dynamically adjusts bitrate based on real-time network conditions rather than using a fixed low bitrate. By continuously monitoring available bandwidth and congestion indicators, the system maintains the highest possible audio quality that the network can support, avoiding unnecessary quality degradation while still preventing congestion.
Solution Approach 2:
The patent changes the bitrate parameter adaptively based on measured network conditions. Instead of selecting a static bitrate, the system modifies this critical parameter in response to changing network state, allowing optimal audio quality when bandwidth is available and reduced quality only when necessary to maintain network stability.
3Measurement precision
If network conditions are monitored continuously to optimize bitrate, then audio quality is optimized, but system complexity increases
Solution Approach 1:
The receiver performs self-service by autonomously measuring network conditions (one-way delay, available bandwidth) and generating bitrate recommendations without requiring complex external control systems. The existing VoIP infrastructure is leveraged for measurements, and standard feedback channels are used for communication, minimizing additional system complexity while achieving optimization.
4Speed
If bandwidth estimation is performed frequently to capture changing network conditions, then responsiveness to network changes is improved, but processing overhead increases
Solution Approach 1:
The system performs bandwidth estimation periodically at defined intervals rather than continuously, balancing responsiveness with processing efficiency. This periodic measurement approach captures significant network condition changes while avoiding the excessive processing overhead of continuous monitoring, achieving a practical compromise for real-time VoIP applications.
Data Source
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AI summary
A computerized method of optimizing audio quality in a voice stream between a sender and a receiver VoIP applications, comprising: defining by the receiver time intervals; determining by the receiver at the end of each time interval whether congestion exists, by calculating (i) one-way-delay and (ii) trend, using double-exponential smoothing; estimating by the receiver a bandwidth available to the sender based on said calculation; sending said estimated bandwidth by the receiver to the sender; and using by the sender said bandwidth estimate as maximum allowed send rate.