RTMP vs SRT: which live streaming protocol should you use?

RTMP (Real Time Messaging Protocol) and SRT (Secure Reliable Transport) are both streaming contribution protocols. used to send video from a production source to a receiving infrastructure (usually encoding or switching equipment). They're often confused, but solve different problems.

RTMP is older (developed by Adobe) and assumes reliable network conditions. It works great in studios with stable fiber connections. It fails gracefully under poor network conditions (drops video rather than adapting). RTMP is simple, widely supported, and has low overhead. However, it lacks encryption and doesn't handle bandwidth fluctuation gracefully.

SRT is newer, designed specifically for unreliable networks. It adapts encoding bitrate based on network conditions, retransmits lost packets, and uses forward error correction to prevent complete video loss during momentary network problems. SRT encrypts streams with AES-256. It's more complex but more reliable for challenging network conditions.

At Creative Broadcast Agency, the choice between RTMP and SRT depends on network conditions at the source. For a studio streaming locally over fiber. RTMP is fine and simpler. For a remote speaker broadcasting from their office over corporate WiFi. SRT is safer. For field coverage using 5G bonding. SRT is essential.

The latency difference is notable: RTMP introduces 1-2 frames latency, SRT introduces 2-3 seconds latency due to its adaptive and retransmission mechanisms. For low-latency streaming where every millisecond matters, RTMP is technically lower-latency. For reliability where occasional latency dips are acceptable, SRT's added latency is worthwhile.

Modern streaming workflows often use SRT for contribution (sending content from remote sources to broadcast infrastructure) and RTMP/HLS for delivery (sending to platforms and CDNs). This hybrid approach captures SRT's reliability advantages for contribution while using established delivery formats.