SRT
Encrypted, error-corrected protocol for reliable delivery over unpredictable networks
Overview
What It Is
Real-Time Messaging Protocol (RTMP) is a TCP-based streaming protocol originally developed by Macromedia (now Adobe) for real-time data, audio, and video delivery. While RTMP’s role as a playback protocol ended with Flash’s deprecation, it remains the dominant ingest protocol for live streaming. Virtually every streaming platform — YouTube Live, Twitch, Facebook Live, Instagram Live — accepts RTMP for ingest. RTMP delivers video with 3–5 second latency, uses persistent TCP connections for reliability, and supports RTMPS (RTMP over TLS) for encrypted ingest. OBS, vMix, Wirecast, and CDN delivery.
Decision Guide
When to Choose RTMP
Choose RTMP when:
- You are ingesting live video to a streaming platform — RTMP is accepted by YouTube, Twitch, Facebook, and virtually every live platform.
- Your encoder is OBS, vMix, Wirecast, or similar — all major encoders output RTMP natively.
- You need reliable delivery over TCP — RTMP’s TCP transport guarantees frame delivery without packet loss.
Consider alternatives when:
- You need sub-2-second latency — WebRTC delivers sub-500ms; SRT delivers 0.5–2s.
- You are sending contribution feeds over unreliable networks — SRT has built-in error correction and encryption specifically designed for lossy networks.
- You need a modern, firewall-friendly protocol — SRT (UDP with Rendezvous mode) traverses firewalls more easily.
Reference
Key Characteristics
| Full name | Real-Time Messaging Protocol |
| Developed by | Macromedia (now Adobe), ~2002 |
| Transport layer | TCP (persistent connection) |
| Typical latency | 3–5 seconds (glass-to-glass) |
| Encryption | RTMPS (RTMP over TLS) |
| Error correction | TCP retransmission (reliable delivery) |
| NAT traversal | Moderate — requires port 1935 (or 443 for RTMPS) |
| Max bitrate | Practically limited by TCP throughput (~50 Mbps typical) |
| Open source | Partially documented; no complete open spec |
| Typical use case | Live streaming ingest to any platform; OBS/vMix/Wirecast output |
Implementation
How Qencode Handles RTMP
Qencode accepts RTMP ingest for live streaming workflows. Send your RTMP stream to a Qencode ingest endpoint, and Qencode handles real-time transcoding, adaptive bitrate packaging (HLS/DASH), and delivery to your storage or CDN.
Combine RTMP ingest with Simulcast to re-stream to multiple platforms simultaneously, or enable DVR Recording to archive the live stream.
Comparison
RTMP vs. SRT
RTMP | SRT | |
|---|---|---|
Transport | TCP | UDP (ARQ) |
Latency | 3–5s | 0.5–2s |
Encryption | RTMPS (TLS) | AES-128/256 (built-in) |
Error correction | TCP retransmit | ARQ (configurable) |
Firewall traversal | Port 1935 | Rendezvous mode |
Platform acceptance | Universal | Growing |
Best for | Platform ingest (YouTube, Twitch) | Contribution feeds, remote production |
For a detailed SRT breakdown, see SRT
Calculate cost of RTMP Protocol
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
Re-stream RTMP input to multiple platforms simultaneously
DVR Recording
Archive live RTMP streams for on-demand playback.
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