SRT is a transport protocol for ultra low-latency live video and audio streaming over unreliable networks.
SRT addresses the challenge of delivering high-quality video streams across congested or noisy networks where packet loss and jitter are common. The protocol maintains constant end-to-end latency by dynamically adapting to real-time network conditions between endpoints. It employs AES encryption to protect media payloads and uses Automatic Repeat reQuest as its primary error recovery mechanism, where receivers alert senders to retransmit missing packets. The tool also supports Forward Error Correction and Connection Bonding for additional stream protection and failover capabilities, reducing the need for buffering on the receiver side.
SRT suits organizations building live streaming workflows where sub-second latency and reliability are critical, particularly for contribution and distribution endpoints in video production. The protocol is appropriate for scenarios involving severe packet loss or bandwidth fluctuations where traditional transport methods fail. It works as an open-source implementation with an Internet Draft specification and an established community of users.
The project maintains active development with regular engagement in its community channels and ongoing refinement of its core protocol implementation and supporting tools.
