Voice Data Sectioning for Uninterrupted Internet Calls
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Solution Overview
Problem
Voice communication over non-QoS guaranteed networks, such as the internet, often experiences interruptions due to packet losses, which degrade voice quality and can render conversations unfeasible, especially when data communication is frequent.
Innovation Solution
A voice data transmitting and receiving system that accumulates voice data from speech start to mute detection, transmits it as a section voice file using packet switching, and employs HTTP protocol with added headers for reliable transmission, re-transmits high-loss packets, and divides voice files to maintain continuous communication.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Speed
If voice data is transmitted using RTP with UDP protocol for real-time communication, then real-time property is improved, but packet losses occur leading to voice interruption
Solution Approach 1:
The transmitting side performs preliminary actions by detecting speech start and mute points, accumulating voice data into complete sections, and adding identification information before transmission. This preparation ensures that received sections can be properly identified and reconstructed without interruption, resolving the contradiction between real-time transmission and voice continuity.
2Reliability
If interpolation is used to cope with packet losses, then voice interruption is reduced, but voice quality deteriorates greatly when packet losses are pronounced
Solution Approach 1:
Voice data is segmented into distinct sections based on speech start and mute detection, with each section containing complete phonetic information. The transmitting side divides continuous voice into discrete, meaningful units and transmits them with identification markers. This segmentation ensures that even if some sections are lost, the remaining sections maintain high voice quality without requiring degradation-prone interpolation.
3Duration of action of moving object
If voice data is transmitted as continuous stream, then real-time communication is maintained, but packet losses cause conversation interruption
Solution Approach 1:
The system implements feedback mechanisms where the transmitting side sends acknowledgment packets to confirm successful transmission of voice sections. The receiving side uses these acknowledgments to determine when to begin reproduction, ensuring that complete sections are received before playback. This feedback loop maintains continuous communication while ensuring conversation feasibility even in the presence of packet losses.
Data Source
AI summary
Voice data from speech start till mute are transmitted as a section voice file to the opposite side voice data transmitting and receiving system, thus permitting voice data transmission free from voice interruption. By transmitting voice data from the speech start till the mute it is possible to eliminate packet losses in the data transmission channel and perform voice communication in the same voice quality as in the transmitting side. Besides, because of the use of HTTP communication port it is possible to obtain voice communication exceeding the rooter without any special setting.