gRPC Performance for Audio and Voice Streaming Applications: A Comparative Analysis with REST API and WebSockets
https://doi.org/10.46610/IJEITSEC.2025.v001i02.005
DOI:
https://doi.org/10.46610/IJEITSEC.2025.v001i02.005Keywords:
Audio streaming, gRPC, HTTP/2, Latency, Microservices, Performance analysis, Protocol Buffers, Real-time communication, REST API, Throughput, Voice streaming, WebSocketAbstract
Real-time audio and voice streaming applications demand low-latency, high-throughput communication protocols. This study presents a comprehensive comparative analysis of three prominent communication protocols: gRPC, REST API, and WebSockets, evaluating their performance in the context of audio and voice streaming applications. Through systematic analysis of latency, throughput, bandwidth consumption, and scalability characteristics, we demonstrate that gRPC significantly outperforms traditional REST API implementations, achieving up to 2.5 times higher throughput and 50–70% lower latency. WebSockets, while effective for real-time bidirectional communication, exhibit higher overhead in CPU utilization compared to gRPC’s HTTP/2 multiplexing capabilities. Our findings indicate that gRPC’s protocol buffer serialization and HTTP/2 multiplexing enable superior performance for bandwidth-constrained environments and low-latency requirements. However, WebSockets remain advantageous for browser-based clients where binary protocol support is limited. This study provides empirical evidence and architectural recommendations for selecting appropriate protocols based on application requirements, network constraints, and scalability needs. Results suggest gRPC is optimal for high-performance audio streaming in microservices architectures, while WebSockets are preferable for browser-based voice applications requiring real-time bidirectional communication. The findings contribute to improving design decisions in modern voice communication systems, IoT applications, and distributed real-time streaming platforms.
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