Voice Encoding via RTCP Feedback for Network Adaptation
Find Innovative SolutionsGenerate Solutions
Solution Overview
Problem
Voice communication on mobile terminals is susceptible to network status fluctuations, leading to poor voice quality due to issues like network delay, packet loss, and jitter, which existing technologies fail to adequately address.
Innovation Solution
A method and apparatus that utilize Real-Time Transport Control Protocol (RTCP) packets to gather network status information, adjust Quality of Service (QoS) parameters, and encode voice data accordingly, ensuring optimal transmission parameters such as packet group length, redundancy packets, and coding rate based on comprehensive network statistics.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If voice data is transmitted over mobile networks, then voice communication is enabled, but voice quality deteriorates due to network delay, packet loss, and jitter
Solution Approach 1:
The system collects network status information from RTCP packets and uses this feedback to dynamically adjust encoding parameters. The encoder monitors network conditions (delay, jitter, packet loss) and adapts its operation accordingly, creating a closed-loop control system that improves voice quality under varying network conditions.
Solution Approach 2:
The encoding parameters are made dynamic rather than fixed. The system adjusts packet group length, redundancy packets, and coding rate in real-time based on current network status, allowing the voice transmission system to adapt to changing network conditions and maintain optimal voice quality.
2Reliability
If network status information is collected and QoS parameters are adjusted in real-time, then voice quality improves, but system complexity increases
Solution Approach 1:
The system changes encoding parameters (packet group length, redundancy packets, coding rate) based on network conditions. By adjusting these parameters dynamically, the system maintains voice quality without requiring a complete redesign of the encoding architecture, thus managing complexity while improving performance.
3Reliability
If encoding parameters are adjusted frequently to adapt to network changes, then voice quality is maintained, but processing time and computational overhead increase
Solution Approach 1:
The system adjusts encoding parameters periodically based on collected network status information from RTCP packets. Rather than continuous adjustment, the system uses periodic updates at appropriate intervals, balancing the need for adaptation with the computational overhead and time required for parameter changes.
Data Source
AI summary
This application relates to a voice encoding and sending method performed at a mobile terminal that is communicatively connected to a plurality of mobile terminals via a computer server. The method includes sending voice data to the plurality of mobile terminals; receiving, from one or more of the plurality of mobile terminals, one or more paths of Real-Time Transport Control Protocol (RTCP) packets based on the voice data received by the one or more mobile terminals; parsing the one or more paths of RTCP packets and generating current network status information from the RTCP packets; adjusting a quality of service (QoS) parameter according to the current network status information; and encoding and sending new voice data according to the adjusted QoS parameter.