RTMP vs SRT
for Live Streaming
RTMP and SRT are the two main protocols streamers use to send live video. RTMP has been the default for over a decade. SRT was built to solve the problems RTMP struggles with: packet loss, unstable connections, and unreliable networks. This page covers what each protocol does in practice, when RTMP is fine, and when SRT is the better choice.
The short answer
- Use RTMP for stable desktop or studio setups with excellent network conditions.
- Use SRT for mobile, IRL, unstable Wi-Fi, or long sessions where drops are costly.
- Use SRT in, RTMP out when streaming through a cloud streaming engine. You get SRT's resilience on the upload and RTMP's compatibility on the platform side.
What RTMP and SRT are
Both are transport protocols for sending live video from a streaming software or an encoder to a server or platform. They solve the same core problem differently.
RTMP (Real-Time Messaging Protocol)
Originally developed for Flash. RTMP runs over TCP, which means the protocol guarantees every piece of data arrives in order. On a stable network this works well. On an unstable connection, that guarantee becomes a problem: when data is lost in transit, the entire stream pauses while the missing piece is re-sent. Viewers see buffering, frozen frames, or the stream drops entirely.
RTMP is the most widely supported ingest protocol. Pretty much every streaming platform accepts it. It works reliably on wired connections and stable Wi-Fi.
SRT (Secure Reliable Transport)
Developed and open-sourced in 2017. SRT takes a different approach: instead of pausing the whole stream when data goes missing, it only re-sends the specific pieces that were lost, within a time budget you configure (the latency setting). If a lost piece can't be recovered in time, the stream continues rather than freezing. The result is a stream that stays watchable even when the network is rough.
SRT was designed for sending live video over unpredictable internet connections, where dropped data and uneven delivery speeds are normal. It handles the conditions where RTMP is more likely to stall or disconnect.
Key differences
The practical differences that affect your stream quality and reliability.
| RTMP | SRT | |
|---|---|---|
| Transport | TCP | UDP with targeted re-sends |
| When data is lost | Stream pauses until the missing data is re-sent | Re-sends only what was lost, skips if too late to recover |
| Network tolerance | Low. Sensitive to unstable connections | High. Designed for unreliable networks |
| Latency | Low on stable networks (~1-2s end-to-end) | Configurable (typically 1-4s, trades latency for resilience) |
| Encryption | Available with RTMPS | AES-128/256 built-in |
| Platform support | Nearly universal (Twitch, YouTube, Kick, etc.) | Limited support. Usually sent through a relay server |
| Setup complexity | Paste a URL and stream key. Done | Configure latency in a URL parameter |
How they actually behave
Technical specs only tell part of the story. Here is what happens in real streaming conditions.
RTMP in practice
On a stable wired connection, RTMP is fast and predictable. You paste a URL and stream key into OBS, hit Start Streaming, and it works. There is nothing to tune. The protocol has been the backbone of live streaming for over a decade, and that maturity means broad compatibility with most encoders and platforms.
The problems appear when the network gets unreliable. Because RTMP requires every piece of data to arrive in order, any lost data triggers a pause while the missing piece is re-sent. On a cellular connection where 2-5% of data goes missing, this translates to frequent micro-stalls. Viewers see buffering, frozen frames, or the stream drops entirely. The encoder often can't recover gracefully and has to reconnect, creating a visible gap in the broadcast.
This isn't a bug in RTMP. The underlying protocol (TCP) is doing exactly what it's designed to do: guaranteeing delivery, even if that means pausing. For live video where a brief visual glitch is acceptable but a 5-second freeze is not, that tradeoff is not ideal.
SRT in practice
SRT handles the same scenario differently. It detects which pieces of data were lost and re-sends only those, within a time budget you configure (the latency setting). If the missing data can be recovered in time, viewers see no glitches. If it can't, the stream skips ahead rather than freezing. The result is a stream that stays live and watchable in conditions where RTMP is more likely to stall or disconnect.
The latency setting is the key tradeoff. Set it too low and SRT doesn't have time to recover lost packets. Set it too high and the delay between your camera and your viewers grows. For most streamers, 1-2 seconds is a practical range for moderately unstable connections. For IRL streaming on cellular, 3-4 seconds gives SRT more headroom to handle brief dead zones.
In most IRL streaming apps, the latency can be adjusted in the settings. In OBS, the latency setting goes in the server URL as query parameters (e.g. srt://host:port?latency=2000000). This isn't hard once you know the syntax, but it's a steeper learning curve than pasting an RTMP URL and stream key. For detailed configuration, see the SRT latency documentation.
When RTMP is the right choice
RTMP still works well in controlled environments where the network is reliable.
Stable wired connections
Desktop streaming over ethernet with consistent upload bandwidth. If your connection doesn't drop packets, SRT's recovery mechanisms don't add value.
Studio setups
Fixed-location broadcasts with a dedicated internet line. The network is a solved problem, so the simplest protocol is good.
Simple workflows
Direct-to-platform streaming where you want the least configuration. Paste the stream key, start broadcasting. No URL parameters to think about.
Short sessions
Brief streams where the probability of a network disruption is low. A 30-minute desktop session on stable Wi-Fi rarely needs SRT.
When SRT is the better choice
Any situation where your network is unpredictable, SRT gives you a significant advantage.
Mobile and IRL streaming
Cellular networks regularly drop data, hand off between towers, and vary in speed. SRT handles all of these while RTMP struggles.
Unstable or congested networks
Event venues, shared Wi-Fi, public networks, or any connection where you can't guarantee quality. SRT degrades gracefully instead of dropping.
Long broadcasts
The longer you stream, the higher the probability of a network hiccup. Over a 6-8 hour stream, even "stable" connections have moments. SRT recovers from these transparently.
Remote production
Sending video between locations over the public internet (field to studio, venue to control room). SRT was built for this use case.
Multi-camera setups over network
When cameras connect over Wi-Fi or a shared network, data loss between devices is common. SRT keeps each feed stable.
Anywhere reliability matters
If a disconnect costs you viewers, ad revenue, or a production. SRT reduces the odds of that happening on any connection type.
Tradeoffs to consider
SRT is the stronger protocol for many streaming scenarios, but it comes with some tradeoffs worth understanding.
Latency is a dial, not a freebie
SRT's resilience comes from its latency buffer. More buffer means more recovery time for lost packets, but also more delay. A 3-second SRT latency setting means your viewers are 3 seconds behind your camera (plus platform-side buffering). For interactive content where chat responsiveness matters, this tradeoff is worth thinking about. For most content, 2-3 seconds of additional delay is okay for viewers.
Configuration is less obvious
RTMP setup is paste-and-go. SRT might require setting parameters in the server URL, namely latency and stream ID. Tools like OBS don't expose these in the UI, so you're editing a URL string directly. It's not difficult, but it's different from what most streamers are used to, and getting the latency value right for your network takes some experimentation.
Platform compatibility gap
Twitch, YouTube, and Kick don't accept SRT ingest directly. You need an intermediary: a relay server or cloud streaming engine that accepts your SRT input and delivers RTMP to platforms. This adds a component to your setup, but also adds benefits like disconnect protection, failover, and multistreaming.
Recommended setups
Practical guidance based on what you're streaming and where.
IRL / Mobile
Use SRT
Cellular networks regularly lose data and deliver it unevenly. SRT keeps the stream watchable through tower handoffs, dead zones, and congestion. Set latency to at least 1-2 sec depending on your coverage. RTMP is more likely to stall or drop in the same conditions.
Studio / Desktop
Either works. SRT recommended
On a stable wired connection, both protocols perform similarly. SRT still gives you a safety net for the occasional network blip. If you're streaming through a relay server anyway, there's little reason not to use SRT. If you stream direct to platform, RTMP is simpler.
Hybrid workflows
SRT for input, RTMP for output
The most common production setup: send SRT from your encoder to a cloud streaming engine, which handles reliability and failover, then delivers RTMP to each output platform over datacenter-grade network. You get SRT's resilience on the upload side and RTMP's compatibility on the delivery side.
SRTLA: SRT with link aggregation
SRTLA extends SRT by bonding multiple network connections into a single stream. If SRT makes one connection more resilient, SRTLA makes multiple connections work together.
With SRTLA, your encoder splits the SRT stream across two or more networks, for example a 5G mobile connection plus Wi-Fi. The receiving server reassembles the packets into a single stream. If one connection drops, the others continue delivering data without interruption.
This is particularly useful for IRL streaming where a single cellular connection can degrade at any time. Mobile apps like Moblin and IRL Pro support SRTLA natively. On the server side, the SRTLA endpoint reassembles the bonded connections into a standard SRT stream for processing.
SRTLA is not a replacement for SRT. It builds on top of it. SRTLA adds the multi-path layer underneath and requires a bit more setup and latency to work effectively.
How Streamrun fits into this
Streamrun accepts all RTMP, SRT, and SRTLA as input protocols and delivers to all platforms. It sits in the cloud between your streaming device and your output destinations.
- Protocol flexibility: send RTMP, SRT, or SRTLA from any encoder. Switch between them without ending your broadcast
- SRTLA support: bond multiple cellular connections from apps like Moblin and IRL Pro for maximum upload reliability
- Disconnect protection: if your encoder drops (regardless of protocol), Streamrun keeps the stream live on platforms while you reconnect
- Failover: configure a backup input or failover video that activates automatically when the primary feed drops
RTMP vs SRT: common questions
Does SRT replace RTMP?
Not entirely. SRT is a better transport protocol for unreliable networks, but RTMP remains the dominant ingest protocol across streaming platforms, encoders, and software. Many streamers send SRT to a relay server or streaming engine, which then delivers RTMP to platforms like Twitch, YouTube, and Kick. SRT replaces RTMP on the upload side where network conditions matter most, but RTMP still handles the last mile to platforms.
Is SRT always better than RTMP?
No. On a stable wired connection with low packet loss, RTMP and SRT perform nearly identically. SRT adds value when the network is unreliable: cellular connections, congested Wi-Fi, long-distance links, or any environment where packet loss is common. If you stream from a desktop on ethernet and never experience drops, RTMP works fine. If you ever stream on mobile or unstable networks, SRT is measurably better.
What SRT latency setting should I use?
Start with 1000-1500ms for a wired or stable Wi-Fi connection. For cellular or unstable networks, increase to 2000-4000ms. Higher latency gives SRT more room to retransmit lost packets before they are needed for playback. Lower latency reduces the delay but increases the chance of visible artifacts under packet loss. The right value depends on your network conditions. Test and adjust.
Can I switch between RTMP and SRT without ending my stream?
If you stream through a relay server or cloud streaming engine like Streamrun, yes. The server maintains the connection to your output platforms regardless of how you send your input. You can disconnect your RTMP input and reconnect with SRT (or vice versa) and your viewers see no interruption.
Does OBS support SRT?
Yes. OBS has built-in SRT support. You select "Custom" as the service, set the server to an srt:// URL, and configure parameters like latency in the URL query string. There is no dedicated latency field in the OBS UI for SRT, so parameters go directly in the server URL (e.g. srt://host:port?latency=2000000). Latency is specified in microseconds in the URL.
What is the difference between SRT and SRTLA?
SRT is the base protocol that handles packet loss recovery over a single network connection. SRTLA (SRT Link Aggregation) bonds multiple network connections (e.g. two SIM cards, or cellular plus Wi-Fi) into a single bonded stream. SRTLA gives you more total bandwidth and redundancy, which is especially useful for IRL streaming where any single connection can drop.
Which streaming platforms accept SRT directly?
Most major platforms (Twitch, YouTube, Kick) do not accept SRT directly from end users. To use SRT, you send your stream via SRT to a relay server or cloud streaming engine, which then delivers it to platforms over RTMP. This is standard practice and adds the benefit of disconnect protection and failover.
Related
IRL Streaming: The Complete Setup Guide
Protocol choice matters most for IRL. Full guide to gear, apps, and configurations for streaming from anywhere.
How to Stream Without Disconnects
Why live streams drop at a systems level and what actually works to prevent disconnects on any network.
Live Stream Failover
Automatically switch to a backup source when your primary feed drops. Works with both RTMP and SRT inputs.
Input Stream Documentation
Set up your input stream with RTMP, SRT, or SRTLA. Includes SRT latency configuration and URL parameter reference.
Stream with the protocol that fits your setup
Streamrun accepts RTMP, SRT, and SRTLA. Switch between them without dropping your broadcast.