RTMP, WebRTC, and SRT: Technical Differences & Use Cases

The comparison is organized under three main sections → RTMP, WebRTC, and SRT → each with clearly defined subheadings, detailed protocol-specific explanations, and relevant code snippets.

RTMP (Real-Time Messaging Protocol)

Protocol Behavior and Transport

RTMP operates over TCP and establishes a persistent socket between the client (publisher) and server (media gateway). It uses its own handshake protocol followed by chunked data transmission. The transport is unidirectional (from encoder to server) for media contribution to CDNs.

Latency and Reliability

Since RTMP relies on TCP, it inherits TCP"s retransmission behavior. This ensures delivery of every byte but results in increased latency during congestion. It typically introduces 2"5 seconds of end-to-end latency.

Encryption and Security

RTMP supports TLS via RTMPS. Enabling RTMPS requires TLS termination at the media server or reverse proxy. Many legacy setups still use unencrypted RTMP.

NAT Traversal

RTMP does not support NAT traversal mechanisms. It requires an open TCP port on the server, making it impractical for peer-to-peer workflows behind NATs or firewalls.

Use Case and Deployment

RTMP is commonly used to ingest live video into platforms such as YouTube, Twitch, or Facebook Live.

Example: Streaming to YouTube with FFmpeg

ffmpeg -re -i input.mp4 -c:v libx264 -preset veryfast -f flv rtmp://a.rtmp.youtube.com/live2/your_stream_key

This command reads input in real-time (-re), encodes video with x264, and streams it using the FLV container over RTMP.

WebRTC (Web Real-Time Communication)

Protocol Behavior and Transport

WebRTC is a peer-to-peer communication protocol based on UDP. It uses ICE for peer discovery, STUN/TURN for NAT traversal, DTLS for secure key negotiation, and SRTP for encrypted media transport. It is inherently bidirectional and low-latency, supporting both media and data channels.

Latency and Reliability

WebRTC prioritizes latency over reliability. Instead of retransmitting lost frames, it skips them to maintain real-time constraints. This results in latencies typically under 500 milliseconds.

Encryption and Security

All WebRTC sessions are encrypted end-to-end using DTLS-SRTP. Encryption is mandatory by the protocol spec. Insecure transmission is not allowed, making it compliant with privacy-sensitive applications.

NAT Traversal

WebRTC includes complete NAT traversal through the ICE framework, and falls back to TURN relays if direct connectivity cannot be established.

Use Case and Deployment

WebRTC is optimal for real-time communications such as video conferencing, remote monitoring, or browser-based broadcasting. Unlike RTMP or SRT, WebRTC requires signaling via a separate channel and custom implementation for stream setup.

Example: Streaming via GStreamer with WebRTC

gst-launch-1.0 -v webrtcbin bundle-policy=max-bundle stun-server=stun://stun.l.google.com:19302 \
videotestsrc ! videoconvert ! vp8enc ! rtpvp8pay ! webrtcbin. \
audiotestsrc ! audioconvert ! audioresample ! opusenc ! rtpopuspay ! webrtcbin.

This sets up a test video and audio stream using GStreamer and pipes it through webrtcbin, initiating ICE/STUN negotiation.

SRT (Secure Reliable Transport)

Protocol Behavior and Transport

SRT runs over UDP but introduces reliability features typically found in TCP, such as packet retransmission (ARQ), selective acknowledgment, and optional Forward Error Correction (FEC). It supports three modes: caller, listener, and rendezvous for both server-initiated and peer-to-peer workflows.

Latency and Reliability

SRT allows configurable latency buffers (e.g., 120ms to 2000ms), letting developers tune for jitter, packet loss, or low-delay streaming. It uses retransmission and FEC to recover from dropped packets without significant delay spikes.

Encryption and Security

Encryption is handled at the transport level using AES. You can configure encryption via passphrase and key length settings.

NAT Traversal

SRT supports rendezvous mode for symmetric NAT traversal, though success depends on both peers initiating the handshake simultaneously and having outbound UDP open.

Use Case and Deployment

SRT is well-suited for contribution feeds in professional workflows to send video from an on-site encoder to a cloud ingest point or studio control room.

Example: Pushing an MPEG-TS Stream via SRT with Encryption

ffmpeg -re -i input.ts -c copy -f mpegts "srt://host_ip:port?mode=caller&latency=200&passphrase=secretkey&pbkeylen=32"

This sends the MPEG-TS stream using SRT in caller mode, with a 200ms latency buffer and AES-256 encryption.

Comparison Table