SRT

Encrypted, error-corrected video delivery over unpredictable networks.

Overview

What It Is

Secure Reliable Transport (SRT) is an open-source protocol developed by Haivision and open-sourced in 2017. SRT uses UDP transport with Automatic Repeat reQuest (ARQ) error correction to deliver video reliably over networks with packet loss, jitter, and fluctuating bandwidth — conditions that break TCP-based protocols like RTMP. SRT includes built-in AES-128/256 encryption, eliminating the need for VPN tunnels for secure contribution feeds. With 0.5–2 second latency and configurable error correction buffers, SRT is the standard for remote production, contribution feeds, and reliable first-mile delivery between broadcast facilities.

Decision Guide

When to Choose SRT

Choose SRT when:
  • You are sending contribution feeds over the public internet — SRT’s ARQ error correction handles packet loss that breaks RTMP.
  • You need built-in encryption without VPN overhead — SRT’s AES encryption eliminates the need for separate VPN tunnels.
  • You are doing remote production across unreliable networks — SRT’s configurable latency buffer lets you trade latency for reliability.
Consider alternatives when:
  • You are ingesting to YouTube, Twitch, or Facebook Live — RTMP is what these platforms accept. SRT adoption is growing but not universal.
  • You need sub-500ms latency for interactive use — WebRTC achieves lower latency at the cost of scalability.
Reference

Key Characteristics

Full nameSecure Reliable Transport
Developed byHaivision (open-sourced 2017)
Transport layerUDP with ARQ (Automatic Repeat reQuest)
Typical latency0.5–2 seconds
EncryptionAES-128 / AES-256 (built-in)
Error correctionARQ-based retransmission, configurable latency buffer
NAT traversalCaller/Listener/Rendezvous modes for various NAT scenarios
Max bitrateNo inherent limit (network-bound)
Open sourceMPL 2.0 (libsrt)
Typical use caseContribution feeds, remote production, reliable first-mile delivery over unpredictable networks
Implementation

How Qencode Handles SRT

Qencode accepts SRT ingest as an alternative to RTMP for live streaming workflows. SRT’s reliability makes it ideal for high-quality contribution feeds that are then transcoded and packaged into HLS/DASH for delivery.

Combine SRT ingest with Simulcast for multi-platform distribution, or enable DVR Recording to archive SRT-ingested streams.

Comparison

SRT vs. RTMP

SRT
RTMP
Transport
UDP (ARQ)
TCP
Latency
0.5–2s
3–5s
Encryption
AES-128/256 (built-in)
RTMPS (TLS)
Error correction
ARQ (configurable buffer)
TCP retransmit
Lossy networks
Designed for packet loss
Breaks under packet loss
Platform acceptance
Growing
Universal
Best for
Contribution feeds, remote production
Platform ingest (YouTube, Twitch)

For a detailed RTMP breakdown, see RTMP

Calculate cost of SRT Protocol

0
Output resolution
480
FPS
30
Transcoding (minutes streamed)
Cost/Minute
SDincludes 480p, 240p
$0.01
HDincludes 1080p, 720p
$0.02
Simulcast / Restream
Cost/GB
$0.04
Explore More

Related Technologies

Simulcast

Distribute SRT-ingested streams to multiple platforms.

DVR Recording

Archive live SRT streams for on-demand playback.

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