Thin Client Audio Streaming via Terminal Server Packet Management
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Solution Overview
Problem
Thin client networks experience latency, packet loss, and out-of-sequence packet arrivals, which negatively impact the performance of real-time audio streaming and digital dictation systems by causing delays and errors in application display and functionality.
Innovation Solution
A system and method that employs a terminal server and client architecture with a device driver and plug-in components, including a codec and recorder/playback module, to efficiently stream audio data by managing packet transmission and reducing network overhead, using a codec like CELP and optimizing bandwidth consumption, and supporting devices like SpeechMike and sound cards.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Ease of operation
If data is transmitted over a network in thin client architecture, then application functionality can be provided to remote terminals, but latency and packet loss occur causing delays and errors in real-time audio streaming
Solution Approach 1:
The system pre-processes audio data on the terminal server before transmission, encoding audio into compressed formats (e.g., CELP) and pre-segmenting data into optimized packets. This preliminary preparation reduces the complexity of real-time network transmission and minimizes latency by having data ready for immediate transmission when requested.
Solution Approach 2:
The patent introduces a terminal server as an intermediary between the audio source and remote terminals. The terminal server acts as a mediator that captures, processes, and redistributes audio data to multiple clients simultaneously. This intermediary architecture centralizes the processing burden on the server, allowing clients to receive pre-processed audio data with reduced network overhead and improved reliability.
2Productivity
If audio data is streamed in real-time over the network, then digital dictation functionality is enabled, but packet loss and out-of-sequence arrivals cause errors in audio playback
Solution Approach 1:
The system segments audio data into discrete, manageable packets with sequential numbering and error-checking information. Each packet is independently transmitted and tracked, allowing the receiving terminal to detect lost or out-of-sequence packets and request retransmission. This segmentation approach transforms a single point of failure into multiple recoverable units, maintaining audio data integrity despite network issues.
Solution Approach 2:
The patent implements feedback mechanisms where receiving terminals monitor incoming audio packets for errors, losses, or sequence violations. When issues are detected, the system requests retransmission of specific problematic packets. This feedback loop ensures that audio data integrity is maintained by correcting transmission errors before they affect playback quality, enabling reliable digital dictation over networks.
3Productivity
If multiple users access the terminal server simultaneously, then network utilization is improved, but server CPU load increases affecting audio processing performance
Solution Approach 1:
The system merges multiple audio streams at the terminal server level, allowing a single audio capture device to serve multiple users simultaneously. The terminal server consolidates audio processing tasks (capture, encoding, compression) into a shared resource that distributes processed audio data to multiple clients. This merging approach eliminates redundant processing on each client device, improving network utilization while managing server CPU load through centralized efficient processing.
Solution Approach 2:
The patent employs parameter changes by dynamically adjusting audio encoding parameters (such as compression ratios, sample rates, and packet sizes) based on network conditions and server load. When server CPU load is high, the system may reduce encoding complexity or increase packet batch sizes to reduce processing overhead. This adaptive parameter adjustment allows the system to maintain multiple user connections while managing server resources efficiently, balancing network utilization with processing performance.
Data Source
AI summary
A system, method and computer readable instructions for controlling streaming audio in real time in a thin client terminal server environment that can be used, for example, to provide support for a digital dictation system. A server runs client software, and at least one terminal runs a driver which communicates with the client software to create a virtual channel over which at least one digital dictation file is transmitted in packetized format between the terminal and the server, such that the client software controls operations performed at the terminal pertaining to the digital dictation file.