Audio Control Server Packet Forwarding for Conference Delay
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Solution Overview
Problem
Existing audio processing systems in videoconferences and conference calls experience significant audio distortion and increased delay due to repeated coding and decoding processes when multiple Multipoint Control Units (MCUs) are involved, and they require extensive audio codec resources, leading to high costs.
Innovation Solution
An audio processing method and system where a control server receives coded audio data from terminals, performs capability negotiation to determine supported protocols, and forwards audio data accordingly, either encapsulating multiple channels into a single logical channel or sending through multiple logical channels, reducing the need for repeated coding and decoding and optimizing resource usage.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If traditional MCU audio processing is used with coding and decoding at each hop, then audio mixing can be achieved, but audio distortion increases significantly
Solution Approach 1:
The patent extracts the audio mixing function from the traditional MCU coding/decoding process. Instead of fully decoding and re-encoding audio at each hop, the system extracts only the necessary audio stream information and forwards it through the network, eliminating redundant coding/decoding operations that cause distortion.
Solution Approach 2:
The patent introduces an intermediary audio forwarding mechanism where coded audio data is directly forwarded between terminals without requiring full decoding and re-encoding at intermediate MCUs. This intermediary approach preserves audio quality while enabling multi-point conferencing.
2Reliability
If traditional MCU audio processing with multiple coding/decoding processes is used, then audio mixing is achieved, but transmission delay increases
Solution Approach 1:
The patent enables continuous audio stream forwarding without interruption for coding/decoding processes. Audio data flows continuously through the network infrastructure, eliminating the stop-start nature of traditional hop-by-hop coding/decoding and significantly reducing transmission delay.
Solution Approach 2:
The patent skips the unnecessary coding/decoding steps at intermediate MCUs by directly forwarding coded audio data. This rushing through of the audio stream without redundant processing dramatically reduces the time required for audio transmission in multi-point conferences.
3Ease of operation
If MCU allocates audio codec ports to each terminal, then audio communication is enabled, but system cost increases
Solution Approach 1:
The patent makes the audio codec ports universal by enabling them to handle multiple terminals simultaneously through the audio stream extraction and forwarding mechanism. Instead of dedicating one port per terminal, the system allows a single port to serve multiple terminals by extracting and forwarding audio streams efficiently.
Solution Approach 2:
The patent merges the audio processing functions across multiple terminals into a unified stream extraction and forwarding approach. Rather than requiring separate dedicated ports for each terminal, the system combines audio handling into a shared infrastructure that reduces the total number of required codec ports.
Data Source
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AI summary
An audio processing method includes: after the terminal accesses the control server, the control server obtains audio capabilities of the terminal through capability negotiation; and the control server forwards the coded audio data to each terminal according to the audio capabilities. An audio processing system and a control server are disclosed. In the embodiments of the present invention, the audio data does not need to undergo an operation of audio coding and decoding every time when the audio data passes through a control server, and the coding and decoding operations required to be performed by the control server are reduced drastically. Especially, in the case that only one control server exists, the audio delay between terminals derive from only network transmission, coding of the sending terminal and decoding of the receiving terminal, and the control server extracts and reassembles the packets of the audio data only. Therefore, the delay is ignorable, the real time of interaction between terminals is enhanced, the control server occupies less audio codec resources, and the costs are reduced.