Live Audio Streaming via UDP Packets for Low-Latency Venues
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Solution Overview
Problem
Existing audio streaming systems experience significant latency and are not optimized for real-time distribution of live audio streams to multiple mobile devices, especially in large venues, due to network variability and inefficient data packet queuing, which limits the number of concurrent clients and adds delay.
Innovation Solution
A system and method utilizing low-latency audio codecs, User Datagram Protocol (UDP) encoding, and edge computing to distribute live audio streams through cellular or entity-dedicated networks, minimizing latency by generating discrete data packets and transmitting them one-at-a-time, and leveraging ancillary processing devices to reduce server load.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If traditional TCP-based audio streaming is used, then audio quality is maintained, but latency increases significantly
Solution Approach 1:
The patent changes the fundamental transmission parameter from TCP (Transmission Control Protocol) to UDP (User Datagram Protocol). This protocol change prioritizes speed and low latency over guaranteed delivery and error correction, enabling real-time audio streaming with minimal delay while maintaining acceptable audio quality through other mechanisms in the system.
Solution Approach 2:
The audio stream is segmented into discrete data packets that are transmitted individually via UDP. Each packet is processed and transmitted independently, allowing for parallel processing and reduced queuing delays compared to traditional buffered streaming approaches.
2Quantity of substance
If multiple concurrent clients are supported, then system capacity increases, but server load and latency increase
Solution Approach 1:
The system segments the client base into groups, with dedicated server instances handling specific client groups. This segmentation prevents any single server from becoming overloaded, maintaining low latency even as total system capacity scales to support many concurrent clients.
Solution Approach 2:
The patent introduces intermediary components including edge computing devices and content delivery network (CDN) nodes that act as intermediaries between the central server and end clients. These intermediaries cache and forward audio packets, reducing the load on central servers and minimizing latency for clients.
3Reliability
If audio data is buffered for quality assurance, then audio quality improves, but latency increases
Solution Approach 1:
The patent extracts the buffering function from the traditional centralized server model and distributes it to edge computing devices and client devices themselves. This extraction allows for minimal local buffering that maintains quality while avoiding the cumulative latency of centralized buffering across multiple network hops.
Solution Approach 2:
The system implements dynamic buffering strategies where buffer sizes and durations are adjusted in real-time based on network conditions, client device capabilities, and audio stream characteristics. This dynamic approach maintains audio quality through adaptive buffering while minimizing latency when network conditions permit.
4Reliability
If network routing is optimized for delivery reliability, then packet delivery improves, but transmission speed decreases
Solution Approach 1:
The patent changes the delivery reliability parameter from guaranteed delivery (TCP) to best-effort delivery (UDP). This parameter change prioritizes transmission speed by eliminating retransmission protocols and error correction overhead, while maintaining acceptable reliability through application-level error handling and audio codec resilience.
Data Source
AI summary
Systems and methods for capturing and distributing livestreams of a live event in real-time to a plurality of mobile computing devices are described. According to some embodiments, the system comprises a memory device configured to store computer-executable instructions that when executed causes a processing device to, upon receipt of the livestream via a first network: generate, in real-time, discrete audio data packets from the livestream; generate a transmission copy of each discrete audio data packet; and transmit the transmission copies one-at-a-time to a provider processing device. The provider processing device is configured to, in accordance with an associated transmission policy, transmit each received transmission copy over a provider network for receipt by the mobile computing devices one-at-a-time. Methods and systems for establishing UDP communication between devices are also