You've got OBS open, a stream key copied, and a server URL pasted in. Hit "Start Streaming" and you're live — simple enough, right? But if you're building a live streaming pipeline at any real scale, the ingest protocol sitting underneath that stream key is one of the most consequential architectural decisions you'll make. Get it wrong, and you pay for it every day in buffering complaints, dropped streams on shaky connections, or a latency floor that makes real-time interaction feel like a satellite phone call.
For most of the last decade, that decision was basically made for you: use RTMP. It was the universal language of live streaming ingest — every encoder spoke it, every CDN accepted it, and the ecosystem just worked. But the landscape has shifted. SRT emerged as a purpose-built challenger for resilient contribution over the public internet, and WebRTC (plus its newer ingest companion, WHIP) has pushed latency into sub-second territory for genuinely interactive use cases.
Today, you have real choices — and the right protocol depends heavily on what you're actually building. This guide breaks down how each protocol works, where each one wins, and how to match the right tool to your specific live streaming use case. Whether you're choosing infrastructure for the first time, migrating off RTMP, or adding a low-latency tier to an existing pipeline, this is the decision framework you need.
How RTMP Works (and Why It Still Matters)
RTMP — the Real-Time Messaging Protocol — was originally designed by Macromedia for Flash Player. Adobe inherited it, and by the mid-2000s it had become the backbone of live streaming ingest worldwide. Flash is long dead, but RTMP never went away, because the encoder ecosystem built around it is enormous.
How it works under the hood: RTMP runs over TCP, using a persistent connection on port 1935 (or port 443 for RTMPS, the TLS-encrypted variant). TCP's reliability guarantees are part of what made RTMP so consistent — every packet is delivered in order or retransmitted. A typical RTMP handshake looks something like this when you're configuring a stream:
Server URL: rtmp://live.example.com/live
Stream Key: your-stream-key-hereFor RTMPS (which you should prefer for any production use):
Server URL: rtmps://live.example.com:443/live
Stream Key: your-stream-key-hereWhere RTMP excels: Universal encoder support is RTMP's single biggest advantage. OBS, Streamlabs, Wirecast, vMix, hardware encoders from Teradek and Haivision, browser-based tools — they all support RTMP. If you need to accept streams from hardware you don't control (your users, your broadcasters, third-party production teams), RTMP is the safe choice. It also performs reliably on stable, low-latency network connections — datacenter-to-datacenter, LAN, and most home broadband setups.
Where RTMP falls short: TCP's reliability guarantee is also its Achilles heel in the real world. When packets are lost on a congested or unstable network, TCP stalls and retransmits — it doesn't skip ahead. Under sustained packet loss, this causes stream freezes and disconnects rather than graceful degradation. There's also a practical latency floor: even under ideal conditions, RTMP ingest combined with a standard HLS delivery pipeline introduces 5–10 seconds of glass-to-glass latency. And Adobe no longer develops the RTMP spec, which means the protocol itself is frozen in time.
When to choose RTMP: When encoder compatibility is your top constraint, or when you're building broadcast-to-VOD workflows where a 5-second latency window is perfectly acceptable. Traditional sports coverage, webinars, church services, gaming streams — these are all solid RTMP use cases. As the Mux team has written about at length in considerations for live streaming ingest, RTMP remains the most practical choice when the "first mile" between your encoder and your ingest server is stable.
How SRT Works (and When It Beats RTMP)
SRT (Secure Reliable Transport) was developed by Haivision and open-sourced in 2017. It's now governed by the SRT Alliance, and some of the world's largest broadcasters — CNN, BBC, Al Jazeera — use it for contribution feeds. If RTMP was built for the Flash era, SRT was built for the public internet era.
How it works under the hood: SRT runs over UDP rather than TCP, which means it doesn't have TCP's built-in reliability mechanism. Instead, it implements ARQ — Automatic Repeat reQuest — at the application layer. ARQ allows SRT to selectively retransmit only the lost packets, without stalling the entire stream. This gives SRT a fundamentally different failure mode than RTMP: under packet loss, it degrades gracefully rather than freezing.
SRT also has native AES-128/256 encryption built in at the protocol level, and a configurable latency buffer that you tune based on your network conditions. That buffer is the key tuning parameter:
# FFmpeg example: ingest via SRT with a 200ms latency buffer
ffmpeg -i input.mp4 \
"srt://live.mux.com:5000?streamid=YOUR_STREAM_KEY&latency=200000"The latency value is in microseconds — 200000 = 200ms. For a clean local network, 120ms might be enough. For a field reporter on a mobile hotspot, you might set it to 2000ms or higher to absorb jitter.
Where SRT excels: SRT is purpose-built for the exact scenario where RTMP struggles most: transmitting high-quality video over public internet connections with unpredictable packet loss and variable latency. Remote live production is the canonical use case. Think of a sports correspondent broadcasting from a stadium over a mobile network, a news team contributing a live feed from the field, or a remote presenter on spotty hotel WiFi. SRT handles all of these scenarios significantly better than RTMP.
The configurable latency buffer also means SRT isn't a fixed-latency protocol — you can tune it for your specific network. On a high-quality connection, SRT can achieve sub-500ms ingest latency. On a difficult WAN link, you trade latency for resilience.
Where SRT falls short: SRT requires UDP ports to be open on firewalls — more IT configuration than RTMP on port 443. Encoder support is good and growing fast (OBS 25+ supports it natively, as does FFmpeg and most modern hardware encoders), but it's not yet the universal default that RTMP is. Configuration is also more complex: you need to understand the latency buffer tuning to get the best results.
When to choose SRT: Remote live production over public internet. Contribution feeds from locations you don't control. Any ingest scenario where you need resilience to packet loss more than you need sub-second latency. The Mux blog post on SRT support going GA goes deeper on the specific reliability gains SRT delivers over real-world connections.
How WebRTC (and WHIP) Works for Live Streaming Ingest
WebRTC was designed by the W3C and IETF for browser-native real-time communication — the same technology that powers Google Meet, Zoom in the browser, and FaceTime on the web. It wasn't originally designed for live streaming ingest, but its latency characteristics make it uniquely valuable for a specific class of use cases.
How it works under the hood: Like SRT, WebRTC uses UDP as its transport layer. But the similarity stops there. WebRTC uses DTLS-SRTP (Datagram Transport Layer Security combined with Secure Real-time Transport Protocol) for encryption and media transport, with built-in congestion control algorithms, jitter buffers, and forward error correction. The result is sub-500ms glass-to-glass latency — genuinely real-time, compared to even the most optimized RTMP or SRT pipelines.
WHIP — the ingest piece that was missing: WebRTC was designed for peer-to-peer connections and lacks a standardized way to push a stream to a server. WHIP (WebRTC-HTTP Ingest Protocol) is an IETF draft standard that adds a simple HTTP-based signaling layer for server ingest. A WHIP ingest request looks roughly like this:
POST /whip/endpoint HTTP/1.1
Host: ingest.example.com
Content-Type: application/sdp
v=0
o=- 0 0 IN IP4 127.0.0.1
...
(SDP offer body)The server responds with an SDP answer and the WebRTC connection is established. WHIP is what allows OBS (from version 30+), browser-based encoder tools, and mobile apps to push WebRTC streams to a server endpoint rather than another browser — which is what makes WebRTC viable for streaming ingest at all. The Mux team's deep dive on WebRTC ingest signaling and WHIP is worth reading for the full technical picture.
Where WebRTC excels: Sub-second latency is WebRTC's defining advantage. For use cases where the gap between something happening and a viewer seeing it is measured in hundreds of milliseconds — not seconds — WebRTC is the only real option. Live auctions where the hammer price appears in real time. Sports betting platforms where odds update with each play. Live commerce where hosts respond to viewer bids. Synchronized trivia games. Real-time Q&A where delay makes conversation awkward. Browser-based encoding is another meaningful advantage: a WebRTC-based broadcast tool requires no software install for the presenter, which dramatically reduces friction in consumer and prosumer use cases.
Where WebRTC falls short: Scaling WebRTC fan-out is genuinely hard. The P2P model it was designed for doesn't translate to one-to-many broadcasting without a Selective Forwarding Unit (SFU) in the middle, and SFUs are complex, stateful, and expensive to operate at scale. Codec support varies across implementations (VP8, VP9, H.264, and AV1 are all in play). And the protocol's inherent complexity — STUN/TURN servers for NAT traversal, ICE candidate negotiation, SDP offer/answer cycles — makes server-side implementation significantly more involved than RTMP or SRT. The state of browser-based live streaming is an evolving story, and the tooling is maturing fast, but it's not yet as turnkey as RTMP.
When to choose WebRTC/WHIP: When genuine real-time interactivity is a core product requirement — not just "low latency," but latency low enough that the delay itself would break the user experience. If your use case is a passive viewing experience (sports broadcast, concert stream, webinar), WebRTC's additional complexity is all cost and no benefit. If your use case requires synchronization with real-world events happening right now, it's the right tool.
Protocol Comparison: RTMP vs. SRT vs. WebRTC
Rather than a table, here's a structured breakdown of the most important dimensions side by side.
Latency: RTMP delivers 3–10 seconds in a typical pipeline. SRT is tunable — from roughly 500ms to several seconds depending on your buffer settings and network conditions. WebRTC achieves sub-500ms, making it the only option for genuine real-time scenarios.
Transport layer: RTMP uses TCP, which provides ordering and reliability guarantees but causes head-of-line blocking under packet loss. SRT uses UDP with ARQ for selective retransmission. WebRTC uses UDP with DTLS-SRTP, plus FEC (Forward Error Correction) and NACK (Negative Acknowledgment) for packet loss handling.
Encoder support: RTMP is supported universally — every encoder, hardware or software, supports it. SRT support is good and growing, covering OBS 25+, FFmpeg, and most modern hardware encoders. WebRTC/WHIP support is still maturing; OBS 30+ added WHIP support, but many encoders don't support it yet.
Encryption: RTMPS wraps RTMP in TLS and is widely supported. SRT has AES-128/256 encryption built natively into the protocol. WebRTC uses DTLS-SRTP encryption by specification — it's always encrypted.
Scalability: RTMP ingest is well-understood and easy to scale. SRT ingest scales similarly. WebRTC fan-out requires SFU infrastructure, which is stateful and complex to scale horizontally.
Best fit: RTMP for broadcast pipelines and VOD workflows with broad encoder support needs. SRT for remote production and contribution over lossy networks. WebRTC/WHIP for sub-second interactive experiences.
A Decision Framework: Choosing Your Ingest Protocol
Step 1 — Define your latency requirement. Start here. If 5-second latency is acceptable (most broadcast, sports, gaming, and event streaming), RTMP is fine. If you need 1–5 seconds with resilience on unstable connections, SRT is the right answer. If you need sub-second latency where delay would break the experience, WebRTC/WHIP is the only path.
Step 2 — Assess your network conditions. Is the ingest path a reliable datacenter link or stable home broadband? RTMP handles it well. Is it an unpredictable public internet connection — a mobile hotspot, a hotel network, a remote location? SRT's packet loss recovery gives you meaningfully better reliability. Is the encoder a browser with no software installation? WebRTC is the only practical option.
Step 3 — Consider your encoder constraints. Are you accepting streams from users with OBS, hardware encoders, or broadcast switchers you don't control? Stick with RTMP, or SRT if encoder support is confirmed. Are you building your own browser-based broadcast tool? WebRTC/WHIP is the natural fit.
Step 4 — Map your audience interaction model. Passive viewing — a sports broadcast, a concert, a company all-hands — has no need for sub-second latency. RTMP or SRT handles this well. If your product requires synchronized real-time interaction — live commerce, wagering, audience participation — that's where WebRTC earns its complexity cost.
A simple decision path: start with latency requirements → if under 500ms, choose WebRTC/WHIP → if 500ms–5s with packet loss concerns, choose SRT → otherwise, choose RTMP. Most production systems evolve along this path over time, starting with RTMP, adding SRT for contribution reliability, and selectively introducing WebRTC for specific interactive moments.
How Mux Supports RTMP, RTMPS, and SRT Ingest
Mux supports RTMP, RTMPS, and SRT ingest natively — no configuration gymnastics required. When you configure your broadcast software for RTMP or RTMPS ingest, you get a stream key and a server URL that works immediately with OBS, Streamlabs, and hardware encoders out of the box. For SRT, the Mux SRT ingest guide walks through the specific connection parameters and latency buffer tuning for production deployments.
One important architectural point: regardless of which ingest protocol you use, Mux transcodes your stream to ABR HLS for delivery. Your choice of ingest protocol doesn't constrain your playback experience — it only affects the "first mile" from encoder to Mux. Mux also supports Low-Latency HLS (LL-HLS) on the delivery side, which reduces playback-side latency to 2–5 seconds and pairs especially well with SRT ingest for end-to-end latency that was difficult to achieve just a few years ago.
Mux does not currently support WHIP ingest. If your use case genuinely requires sub-second WebRTC ingest, you'll need a dedicated WebRTC infrastructure provider or a self-hosted SFU. But for the vast majority of production live streaming use cases — broadcast, events, sports, gaming, remote production — RTMP and SRT with LL-HLS delivery covers the full range of what you need.
Choosing the Right Tool for the Right Use Case
The core tradeoff is a triangle: compatibility (RTMP), resilience (SRT), and latency (WebRTC). No single protocol wins on all three dimensions, and that's not a failure of the ecosystem — it's a reflection of the fact that live streaming use cases genuinely have different requirements.
RTMP's staying power isn't nostalgia; it's the encoder ecosystem. SRT's rise isn't hype; it solves a real problem that RTMP can't — reliable video contribution over the public internet. WebRTC's adoption for streaming ingest isn't overreach; it unlocks interaction models that 5-second latency makes impossible.
Most teams building serious live streaming infrastructure will end up using more than one of these protocols at some point. Start with what your current use case demands. Migrate when your requirements evolve. And when you're ready to build, Mux's live streaming quickstart gets you from zero to a working ingest pipeline in under 15 minutes — with both RTMP and SRT supported on day one.