WebRTC vs SRT vs RTMP: Protocol Comparison

The question comes up on every new project: WebRTC, SRT, or RTMP? The frustrating answer is that all three are reasonable choices depending on what you're actually building. They solve different problems, and picking the wrong one tends to cause the kind of subtle failures that are hard to debug later.

This article is structured around architectural decisions rather than protocol internals. The goal is to help you reason about trade-offs, not memorize specs.

What Each Protocol Was Built to Do

RTMP (Real-Time Messaging Protocol) was designed by Macromedia in the early 2000s to stream Flash content over TCP. It's been obsolete in theory for years. In practice, it's still the default ingest protocol for almost every cloud streaming platform — YouTube, Twitch, Wowza, AWS Elemental, all of them. It works, it's supported everywhere, and its latency of 2–10 seconds is acceptable for most live streaming scenarios.

The problem is that RTMP was never designed for the modern internet. It doesn't handle packet loss well (it's TCP, so retransmits block the stream). It has no built-in encryption. Its Flash origins mean it carries architectural baggage that manifests in odd ways when you push it past its design envelope.

SRT (Secure Reliable Transport) is the protocol Haivision built and open-sourced in 2017, specifically for professional broadcast contribution over unpredictable networks. The core design insight: UDP with a reliability layer on top that lets you tune the trade-off between latency and packet recovery. You set a latency buffer (typically 120–500ms for contribution links), and the protocol uses that buffer to retransmit lost packets before they cause visible artifacts.

SRT also has native encryption (AES-128/256), caller/listener/rendezvous connection modes, and strong support for NAT traversal. It was designed for the use case where you're contributing a high-quality feed from a location with unreliable connectivity — a stadium, a remote site, a mobile truck.

WebRTC started as a browser API for real-time communication — video calls, conferencing, that kind of thing. Sub-second latency is its defining characteristic. It uses DTLS for encryption and SRTP for media, runs over UDP with RTCP feedback loops, and adapts aggressively to network conditions through congestion control algorithms (GCC, REMB).

The trade-off for that sub-second latency: WebRTC's congestion control can be aggressive about dropping quality to maintain timing. Under bad network conditions, it will reduce resolution and frame rate rather than buffer. That's the right call for interactive communication, where staleness is worse than lower quality. It's the wrong call for passive viewing where you'd rather have a 2-second buffer and smooth playback.

When the Choice Actually Matters

Most of the time, the choice is made for you. You're ingesting to a cloud platform? RTMP, because that's what they accept. You're doing a video call? WebRTC, because it runs in the browser. The hard decisions happen at the edges:

Contribution over a cellular or satellite link — SRT wins here. The latency buffer absorbs packet loss without stalling the stream. RTMP on a lossy link will retransmit at the TCP level, causing the entire stream to stall while packets catch up. WebRTC will aggressively reduce quality, which may not be acceptable for broadcast contribution.

Sub-500ms live streaming to a large audience — This is harder than it sounds. WebRTC is designed for point-to-point or small groups. Scaling it to thousands of viewers requires infrastructure (SFUs, CDN with WebRTC support) that most teams don't have. SRT can achieve 120–200ms latency on good networks, and scales better because you can transcode to HLS/DASH at the edge.

Interactive applications with live video — WebRTC. If the user needs to see the result of their action within a second, nothing else comes close.

Legacy system integration — RTMP. If you're connecting to hardware encoders, broadcast equipment, or platforms that haven't updated their ingest, RTMP is still the common language.

The Comparison That Actually Matters

Rather than a spec table, here's how each protocol behaves under the conditions that cause real problems:

Packet loss at 2% — At 2% packet loss, a 1080p RTMP stream starts stuttering because TCP retransmits pile up. SRT at 200ms latency buffer recovers cleanly — the buffer is large enough to retransmit and deliver in order. WebRTC reduces bitrate to stay real-time, so quality drops but the stream continues.

Firewall traversal — RTMP is TCP on port 1935; firewalls often allow it. WebRTC uses STUN/TURN for NAT traversal and typically gets through corporate firewalls but requires a TURN server for the hard cases. SRT is UDP, which many firewalls block by default; you often need explicit rules.

Scaling — RTMP has good CDN support because it's been around forever. SRT scales through relay chains or conversion to HLS/DASH at the edge. WebRTC scaling requires an SFU (Selective Forwarding Unit) like mediasoup or Janus, which is real infrastructure work.

Encoder support — RTMP is supported in every hardware encoder, OBS, and most software encoders. SRT support has grown significantly since 2017 and now covers most broadcast hardware. WebRTC encoding is mostly software — hardware encoders rarely expose WebRTC directly.

A Realistic Architecture Decision

Here's a pattern that shows up often in broadcast-adjacent applications: a contribution chain from field to cloud to viewer.

Field encoder → SRT → cloud transcoder → RTMP or HLS → viewers

The field-to-cloud leg uses SRT because the field connection is unreliable. The cloud-to-viewer leg uses HLS/DASH because that's what CDNs know how to deliver at scale. RTMP shows up if there are legacy ingest points in the chain.

Adding WebRTC usually means adding a separate interactive layer alongside the passive viewing layer — a talkback channel, a return feed for the talent, a director's cut that a producer can watch with low latency.

The mistake is trying to use one protocol for the entire chain. SRT is not great for browser playback. WebRTC is not great for contribution over bad networks. RTMP is not great at anything except compatibility.

What Teams Get Wrong

Treating latency specs as absolute — Every protocol's latency number assumes decent network conditions. A 120ms SRT stream over a lossy 4G connection with a small buffer will stutter. Spec your latency budget with a realistic network model.

Ignoring the encryption gap — RTMP has no native encryption. If you're sending a live stream over the public internet without a VPN or TLS tunnel, that stream is readable by anyone on the path. SRT and WebRTC both encrypt by default.

Underestimating WebRTC infrastructure — The browser API is simple. The infrastructure behind it is not. Running a reliable WebRTC system at scale requires STUN/TURN servers, an SFU, monitoring, and capacity planning. Teams often underestimate this until they've already committed to the approach.

Building around a single protocol when the use case needs two — Broadcast production workflows often need low-latency monitoring (WebRTC or SRT) alongside high-quality delivery (HLS/DASH). These are separate pipelines that solve separate problems.

Where Medialooks Fits

Medialooks' SDK components handle multi-protocol I/O natively — RTMP, SRT, and WebRTC inputs and outputs without switching between different libraries. For teams building applications that span multiple protocols in a single pipeline, that matters more than it sounds. Protocol conversion and format handling are usually where the integration work accumulates.

WebRTC, SRT, and RTMP aren't competing for the same use cases — they mostly occupy different parts of the streaming stack. The decisions get hard at the boundaries: contribution over bad networks (SRT usually wins), low-latency delivery to browsers (WebRTC or SRT-to-HLS depending on scale), and legacy integration (RTMP because there's no alternative).

Match the protocol to the network conditions and the latency requirements, not to what's easiest to implement first.