Ultra-low latency streaming with the SRT protocol
SRT (Secure Reliable Transport) is an open-source video transport protocol designed to deliver high-quality, low-latency video across unpredictable networks, including the public internet. Unlike RTMP, SRT was built from the ground up for modern broadcasting workflows and includes built-in error correction, encryption, and congestion control. It is maintained by the SRT Alliance and supported by hundreds of industry vendors.
SRT achieves glass-to-glass latency as low as approximately 120 milliseconds under optimal network conditions. This makes it ideal for interactive broadcasts, remote interviews, and real-time production scenarios where delay must be minimized. The protocol's Automatic Repeat reQuest (ARQ) mechanism retransmits lost packets without the overhead of TCP, maintaining both low latency and high reliability.
SRT connections use a handshake between two roles:
Choose the mode based on your network topology. In most contribution-feed setups, the ingest server runs as a listener and the field encoder connects as a caller.
StreamDev lets you set the SRT latency buffer anywhere from 20 ms to 8,000 ms. The latency value determines how much buffer the protocol uses for packet retransmission. Lower values reduce delay but leave less room for error recovery; higher values improve reliability on lossy networks at the cost of added delay.
Recommendation: Start with 120–250 ms for local or low-loss networks. For long-distance or unstable connections, increase to 500–2,000 ms for more resilient delivery.
SRT supports AES encryption to protect your stream in transit. StreamDev offers two encryption levels:
Encryption is enabled by setting a passphrase in the stream configuration. Both the sender and receiver must use the same passphrase and encryption key length. Without the correct passphrase, the stream cannot be decoded.
SRT output in StreamDev is designed for professional broadcasting workflows: