Adaptive Voice Packet Buffer Control for VoIP Continuity
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Solution Overview
Problem
Conventional buffer control techniques in packet switched networks fail to adapt smoothly to varying network conditions, leading to interruptions and unnatural sound in VoIP and video transmissions due to inaccurate end-to-end delay measurements and inappropriate buffering delay adjustments.
Innovation Solution
The adaptive buffer control mechanism measures interruptions that would occur without reserve delay and adjusts the buffering delay based on the arrival interval and interruption delay to maintain a continuous playout signal, using a combination of base and reserve delays managed through time scaling.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If conventional fixed delay buffering is used, then packet arrival time variations are reduced to some extent, but the system cannot react to changing network conditions and buffer underflows occur when throughput is consistently low
Solution Approach 1:
The patent implements dynamic buffer delay adjustment by continuously monitoring packet arrival intervals and adapting the buffer delay parameter in real-time. The buffer delay is increased when packet arrival intervals exceed a threshold and decreased when they fall below the threshold, enabling the system to dynamically respond to changing network conditions while maintaining continuous voice output
Solution Approach 2:
The system employs feedback mechanisms by measuring the actual packet arrival intervals and using this information to adjust the buffer delay parameter. The measured arrival intervals are compared against thresholds to determine whether to increase or decrease buffering, creating a closed-loop control system that adapts to network conditions
2Adaptability or versatility
If adaptive buffer control frequently adjusts buffering delay, then response to network conditions improves, but unnecessary fluctuations in playout rate occur causing unnatural sounding speech
Solution Approach 1:
The patent introduces periodic action by requiring a minimum number of consecutive threshold exceedances (counter threshold) before triggering a buffer delay adjustment. This filtering mechanism prevents reactive adjustments to transient network variations and reduces unnecessary fluctuations in playout rate, thereby maintaining natural-sounding speech while still responding to sustained network condition changes
Solution Approach 2:
The system performs preliminary verification by counting consecutive threshold exceedances before executing buffer delay changes. This preliminary action ensures that only sustained network condition changes trigger adjustments, preventing premature or spurious adjustments that would cause unnatural speech variations
3Ease of manufacture
If fixed sampling interval is used for buffer control, then implementation is simplified, but reaction time increases when packets arrive at reduced rate potentially causing speech interruptions
Solution Approach 1:
The patent replaces fixed sampling intervals with dynamic event-driven sampling triggered by packet arrivals. The buffer delay is adjusted based on actual packet arrival intervals rather than predetermined time slots, enabling the system to respond immediately to changes in packet arrival rates without the delay inherent in fixed sampling schemes
4Reliability
If buffer delay is increased to prevent underflows, then continuous output is maintained, but interruptions occur when packets arrive at reduced rate during sampling intervals
Solution Approach 1:
The system uses feedback from actual packet arrival intervals to trigger buffer delay adjustments immediately when needed. By monitoring each packet arrival and comparing its interval against thresholds, the system provides real-time feedback that enables prompt buffer delay adjustments, preventing both underflows and unnecessary delays
Data Source
AI summary
Disclosed is a device, a computer program and a method to receive and buffer data packets that contain information that is representative of time-ordered content, such as a voice signal, that is intended to be presented to a person in a substantially continuous and substantially uniform temporal sequence; to decode the information to obtain samples and to buffer the samples prior to generating a playout signal. The samples are time scaled as a function of packet network conditions to enable changing the play-out rate to provide a substantially continuous output signal when the data packets are received at a rate that differs from a rate at which the data packets are created. The time scaling operation operates with a base delay that is controlled in a positive sense when the data packets are received at a rate that is slower than a rate at which the data packets are created, and a reserve delay that is managed to provide insurance against an interruption should the base delay become negative.