A clear, operator-level comparison of RTMP and SRT for live stream contribution: latency, reliability, security, and when each protocol is the right choice.
RTMP vs SRT: Live Streaming Ingest Protocols Explained
By Sampath Mallidi, CEO of Revidd · Last updated June 2026
RTMP vs SRT is a contribution-protocol decision, not a delivery one. RTMP is the legacy default for sending a live feed from an encoder to a streaming platform: ubiquitous, easy, but built on TCP and weak over unstable networks. SRT is newer, runs on UDP, and is built for reliability, security, and low latency over the public internet. Use RTMP when your destination only accepts RTMP. Use SRT when the network between camera and server is unpredictable.
TL;DR
RTMP = ubiquitous, simple, supported almost everywhere, but unreliable on lossy or long-distance networks and limited to older codecs.
SRT = reliable and secure over bad networks, lower latency, codec-agnostic, but not accepted by every destination yet.
Both are contribution (ingest) protocols. They get your feed to the platform. They are not how viewers watch. Viewers get the stream over HLS or DASH.
Field production, remote sites, and shaky connections favor SRT. Pushing into a platform that only takes RTMP favors RTMP.
A capable streaming platform should accept both so you are not boxed in by the protocol your encoder happens to speak.
What is the difference between RTMP and SRT?
RTMP and SRT are both protocols for getting a live video feed from an encoder or camera into a streaming server. The core difference is the network they were designed for. RTMP runs over TCP and assumes a stable connection. SRT runs over UDP and is engineered to keep a stream stable across the unpredictable public internet.
RTMP (Real-Time Messaging Protocol) was created by Macromedia for Flash. Flash is gone, but RTMP survived as the de facto standard for live contribution because encoders, software, and platforms all spoke it. It is simple to set up and accepted nearly everywhere.
SRT (Secure Reliable Transport) is an open-source protocol developed by Haivision and now maintained by the SRT Alliance. It was built specifically to move broadcast-quality video over lossy networks with low latency and built-in encryption. It detects which packets were lost and re-sends only those within a configurable time budget, so a noisy connection that would stall RTMP can stay live on SRT. According to Haivision's RTMP versus SRT white paper, SRT maintained reliable contribution at higher bitrates and over longer distances where RTMP failed.
What does contribution mean, and how is it different from delivery?
Contribution is the first mile: getting the live feed from the source into your platform. Delivery is the last mile: getting the stream from your platform out to thousands of viewers. RTMP and SRT are contribution protocols. They are not how the audience watches.
This distinction trips up a lot of broadcasters. You contribute in with RTMP or SRT. You deliver out with HLS or DASH, which are segmented HTTP protocols designed to scale to large audiences across devices. If you want the detail on the delivery side, see our explainer on what HLS is and how it works.
A typical live workflow looks like this:
Source: camera, switcher, or encoder.
Contribution (ingest): RTMP or SRT pushes the feed to the platform.
Transcoding: the platform creates multiple bitrate renditions.
Delivery: HLS or DASH streams to viewers over a CDN.
Playback: apps on Roku, Apple TV, Samsung, LG, iOS, Android, and web.
Pick the contribution protocol for the network conditions of the first mile. Pick the delivery protocol for reach and device coverage. They are separate decisions.
RTMP vs SRT: side-by-side comparison
The table below summarizes the practical differences broadcasters care about. Figures for latency and codec support are drawn from Haivision documentation and protocol specifications, not Revidd benchmarks.
Factor | RTMP | SRT |
|---|---|---|
Transport layer | TCP | UDP |
Typical glass-to-glass latency | 2 to 5 seconds | around 1 second |
Behavior on packet loss | Stalls, buffers, or disconnects | Re-sends lost packets within a time budget; skips ahead rather than freezing |
Long-distance / lossy networks | Unreliable above ~2 Mbps | Holds higher bitrates over poor links |
Encryption | None native (relies on RTMPS/TLS) | Built-in AES-128 / AES-256 |
Codec support | Older set; no HEVC, VP9, or AV1 | Codec-agnostic (carries any format) |
Adoption | Near-universal; default for most platforms | Growing fast; not accepted everywhere yet |
Best for | Simple ingest into platforms that require it | Remote production, field feeds, unstable networks |
When should a broadcaster use RTMP?
Use RTMP when your destination requires it or when your network is stable and local. RTMP is still the path of least resistance for a lot of live workflows because almost every encoder, software switcher, and platform supports it out of the box.
Reach for RTMP when:
Your streaming destination only accepts RTMP input.
The connection between encoder and server is stable, local, and low-latency.
You want the simplest possible setup with hardware and software you already own.
You are restreaming to social platforms that default to RTMP ingest.
The weakness is real. RTMP rides on TCP, so a single dropped packet triggers retransmission that backs up the whole stream. On a congested venue Wi-Fi or a long-haul connection between continents, that is when RTMP stutters, buffers, or drops. For a regional station pushing a stable feed from a fixed gallery, none of that matters. For a sports crew on a stadium uplink, it matters a lot.
Setting up live for the first time? If you are a broadcaster or rights holder launching live without an in-house video engineering team, the protocol is only one piece. You also need transcoding, delivery, monetization, and apps on every screen. See how broadcasters launch a live sports streaming platform end to end, or book a Revidd demo to see the full ingest-to-screen path.
When should a broadcaster use SRT?
Use SRT when the network between your source and your platform is unpredictable, when you need lower latency, or when the feed must be encrypted in transit. SRT was built for exactly the conditions RTMP struggles with: remote production, field contribution, and long-distance links over the public internet.
Reach for SRT when:
You are contributing from a remote site, an outside broadcast, or a venue with shaky connectivity.
You need lower latency for interactive or near-live use cases.
The feed crosses long distances or international links.
Security matters and you want encryption built into the transport.
You are sending newer codecs such as HEVC that RTMP cannot carry.
SRT's recovery model is the reason it stays live where RTMP fails. Instead of forcing retransmission of everything in sequence like TCP, it tracks individual lost packets and re-requests only those, within a latency budget you set. If a packet can be recovered in time, the viewer sees nothing wrong. If it cannot, the stream skips ahead instead of freezing. That is why field crews and broadcast engineers have moved to SRT for contribution over networks they do not control.
Revidd accepts SRT-type live streams as an ingest source, alongside RTMP push and other live inputs, so a broadcaster can use the contribution protocol that fits the site rather than the one the platform happens to allow.
Is SRT replacing RTMP?
SRT is steadily taking over the contribution side, but RTMP is not gone. RTMP remains the most widely supported ingest protocol, especially for pushing into consumer platforms, so it stays relevant as a fallback and a compatibility layer. SRT is winning where reliability and latency matter most: remote production, broadcast contribution, and unstable networks.
The realistic position for a broadcaster in 2026 is not "pick one forever." It is "use a platform that handles both." Your encoder, your venue, and your destination determine which protocol fits a given event. The smart move is to keep both options open so the protocol never becomes the thing that limits you.
Neither RTMP nor SRT touches the delivery side. Once your feed is in, the platform transcodes it and delivers to viewers over HLS or DASH. If you are also planning linear channels off a live feed, see how sports rights holders launch a D2C streaming channel across VOD, live, and FAST.
What this means for broadcasters launching live
The protocol you ingest with should never be the bottleneck in your live operation. RTMP vs SRT is a first-mile question that depends on your network, your gear, and your destination. The platform behind it needs to accept whatever your field setup throws at it, transcode it, and deliver it to every screen your audience uses.
Revidd is a plug-and-play OTT and FAST platform built for broadcasters who have a video library and need to go live, on-demand, and linear across every major device without building infrastructure in-house. The platform reaches more than 38 million viewers and a 5.2 million monthly active audience across 15 countries, with broadcast-grade live, EPG, SCTE-35 ad insertion, and Rescue Playlist failover. It accepts RTMP and SRT-type live ingest, transcodes to multiple renditions, and delivers over HLS to native apps on Roku, Apple TV, Android TV, Samsung, LG, Vizio, iOS, Android, and web from a single integration.
If you are a faith network, sports rights holder, regional station, or diaspora channel planning live streaming and you do not want to wire together ingest, transcoding, delivery, and apps yourself, book a Revidd demo and we will walk you through the full path from contribution to screen.
FAQ
Is RTMP or SRT better for live streaming?
Neither is universally better. SRT is better for reliability, latency, and security over unstable or long-distance networks, which is why it has become the preferred contribution protocol for remote production. RTMP is better when you need maximum compatibility, since almost every encoder and platform still accepts it. The right choice depends on your network and your destination.
Are RTMP and SRT used for delivery to viewers?
No. RTMP and SRT are contribution (ingest) protocols that move a live feed from a source to a streaming platform. Viewers receive the stream over delivery protocols such as HLS or DASH, which are designed to scale to large audiences across devices. Contribution and delivery are separate stages of the workflow.
Does SRT have lower latency than RTMP?
Yes. SRT typically achieves around one second of glass-to-glass latency over the public internet, while RTMP usually lands between two and five seconds, per Haivision documentation. SRT also degrades more gracefully under packet loss, recovering lost packets within a configurable time budget instead of stalling.
Why do platforms still use RTMP if SRT is better?
RTMP remains the default because of universal support. Encoders, software switchers, and consumer platforms have spoken RTMP for years, so it is the lowest-friction way to push a feed. SRT support is growing quickly, but until it is accepted everywhere, RTMP stays useful as a compatible fallback.
Can one streaming platform accept both RTMP and SRT?
Yes. A capable OTT and live platform should accept multiple ingest types so the contribution protocol is chosen by your field conditions, not dictated by the platform. Revidd accepts RTMP push and SRT-type live streams as ingest sources, then transcodes and delivers the result over HLS to apps on every major device.
