What is SRT streaming? Secure Reliable Transport explained.

SRT stands for Secure Reliable Transport. It's an open-source video transport protocol that moves live video over the public internet with broadcast-grade reliability, sub-second latency, and end-to-end encryption. Originally developed by Haivision and released as open source in 2017, SRT has become the default transport layer for professional live video , replacing both satellite contribution and RTMP streaming in most production workflows.

Creative Broadcast Agency is a member of the SRT Alliance , the consortium of over 600 companies advancing SRT as an open standard. We use SRT on every production we deliver, from corporate events in Dubai to the Esports World Cup in Riyadh. This isn't a theoretical explainer , it's how we use SRT in practice every week.

What Does SRT Stand For?

SRT stands for Secure Reliable Transport. It is an open-source video transport protocol designed to send live video across the public internet with broadcast-grade quality, sub-second latency, and AES encryption on every stream. Developed by Haivision and released as open source in 2017, SRT is now maintained by the SRT Alliance , a consortium of over 600 broadcast, streaming, and technology companies.

In practical terms, SRT is the protocol that replaced satellite uplinks and RTMP for moving live video from point A to point B. If you are researching what is SRT streaming for your own production workflow, the short answer is: it is the current industry standard for professional live video transport, used by broadcasters, esports leagues, corporate event producers, and government agencies worldwide.

For a full glossary definition, see our SRT (Secure Reliable Transport) glossary entry.

How SRT Works , The Technical Fundamentals

To understand why SRT matters, you need to understand the problem it solves.

The public internet is unreliable. Data packets get lost, arrive out of order, or show up late. Traditional streaming protocols handle this in one of two ways: RTMP accepts the quality loss (dropped frames, artefacts, frozen video), and HLS adds significant delay (15-45 seconds) to buffer against network problems. Neither is acceptable for professional broadcast.

SRT takes a different approach. It solves three problems simultaneously.

Automatic Repeat reQuest (ARQ)

When SRT detects a lost packet , which happens constantly on any internet connection , it automatically requests retransmission from the sender. This happens within a configurable latency window, typically 60-250 milliseconds. The decoder receives the retransmitted packet before it even notices anything was missing. The result is broadcast-quality video over connections that would destroy an RTMP stream.

This is fundamentally different from Forward Error Correction (FEC), which sends redundant data regardless of whether errors occur. ARQ only retransmits what's actually lost, making it more bandwidth-efficient.

Configurable Latency Buffer

SRT lets you set exactly how much delay the protocol has to work with for error correction. Lower latency means less time to recover lost packets. Higher latency means more resilience against network problems.

For live event contribution feeds (camera to MCR), we typically set SRT latency to 120-250 milliseconds. This gives enough headroom for error correction while keeping the total glass-to-glass latency under one second. For interactive applications , live auctions, sports betting feeds, esports , we can drop to 60-80ms and accept that error correction will be less aggressive.

This configurability doesn't exist in RTMP. With RTMP, you get whatever the network gives you.

AES Encryption

Every SRT stream is encrypted with AES-128 or AES-256. This isn't optional , it's built into the protocol. There is no unencrypted SRT mode. For corporate streaming (board meetings, financial results), government broadcasts, or any content that can't leak, this is critical. The encryption adds negligible overhead , we use AES-256 on every feed by default.

SRT vs RTMP , Why RTMP Is Dead for Professional Use

RTMP (Real-Time Messaging Protocol) was developed by Adobe in the early 2000s for Flash Player. It's still accepted by most streaming platforms for backward compatibility, but it has fundamental limitations that make it unsuitable for professional production.

No error correction. When RTMP packets are lost in transit , which happens regularly on busy venue networks , the stream shows visible artefacts, frozen frames, or drops entirely. There's no recovery mechanism. SRT recovers lost packets automatically within the latency window.

No encryption. RTMP has no built-in encryption. Anyone on the same network can intercept an RTMP stream. SRT encrypts every stream with AES by default.

No modern codec support. RTMP was designed for H.264 inside a Flash container. It doesn't natively support H.265 (HEVC) or AV1 , the codecs that deliver the same quality at half the bitrate. SRT is codec-agnostic , it transports whatever the encoder outputs.

No adaptive behaviour. RTMP has no mechanism to monitor network conditions or adapt in real time. SRT continuously tracks round-trip time, packet loss, and available bandwidth, adjusting its retransmission behaviour accordingly.

Here is a direct comparison of the two protocols. For a deeper breakdown, see our dedicated RTMP vs SRT comparison.

Feature	SRT	RTMP
Latency	60-250 ms (configurable)	1-5 seconds typical
Error correction	ARQ automatic retransmission	None , lost packets cause artefacts or drops
Encryption	AES-128 / AES-256 built in	None built in (requires external TLS wrapper)
Codec support	Codec-agnostic (H.264, H.265, AV1, etc.)	H.264 only (natively)
Network adaptability	Real-time RTT, loss, and bandwidth monitoring	No adaptive behaviour
Open source	Yes , maintained by SRT Alliance	Proprietary (Adobe), partially reverse-engineered
Firewall traversal	Caller/listener/rendezvous modes	Requires specific port forwarding
Use case in 2026	Industry standard for contribution and transport	Legacy ingest to some platforms only

The bottom line: if your streaming partner is using RTMP as their transport protocol in 2026, they're using technology that was obsolete years ago.

SRT vs HLS , Transport vs Delivery

HLS (HTTP Live Streaming) and SRT serve fundamentally different purposes, but they are often confused because both involve moving live video over the internet.

SRT is a transport protocol. It moves a single live video stream from point A to point B , camera to MCR, venue to broadcast partner, encoder to decoder. It is optimised for low latency, error correction, and point-to-point reliability.

HLS is a delivery protocol. It packages video into short segments (typically 2-6 seconds) served over standard HTTP, designed for scalable distribution to thousands or millions of viewers via CDN. HLS prioritises compatibility and scalability over latency.

Feature	SRT	HLS
Purpose	Point-to-point transport	Scalable audience delivery
Typical latency	60-250 ms	15-45 seconds (standard), 2-6 seconds (LL-HLS)
Error handling	ARQ retransmission in real time	Segment-based retry (viewer sees buffering)
Encryption	AES built in	AES-128 via HLS encryption spec
Scalability	Point-to-point or limited multicast	Millions of concurrent viewers via CDN
Codec support	Codec-agnostic	H.264, H.265 (device-dependent)
Use case	Contribution, remote production, backhaul	Final delivery to audience

In a typical CBA production, both protocols are in play. SRT handles every contribution and backhaul feed , venue to MCR, MCR to broadcast partners. HLS (or DASH) handles the final delivery to the public audience via YouTube, social platforms, or a branded player on the client's website. They are complementary, not competing.

For productions that need low-latency streaming to the end viewer , live auctions, esports betting feeds, interactive Q&A , we use WebRTC or LL-HLS for the last mile, with SRT still handling the transport pipeline behind it.

SRT vs RIST

RIST (Reliable Internet Stream Transport) solves similar problems to SRT , error correction, encryption, and low-latency transport over the public internet. Both are open standards backed by industry consortiums.

In practice, SRT has significantly broader hardware and software support. Every major encoder, decoder, switcher, and cloud platform supports SRT natively. Most equipment that supports RIST also supports SRT, but the reverse isn't always true. We default to SRT for all production workflows and only use RIST when a specific broadcast partner requires it.

SRT vs NDI

NDI (Network Device Interface) is designed for local network transport , connecting cameras, switchers, and graphics engines within a single venue or studio. It's excellent for low-latency, high-quality video over a managed LAN, but it's not designed for transport over the public internet.

In our production workflow, NDI and SRT serve different purposes. We use NDI inside the venue for local device connectivity, and SRT for anything that needs to cross the internet , contribution feeds to our MCR, delivery to broadcast partners, or remote production links.

SRT vs WebRTC

WebRTC is designed for real-time communication , video calls, interactive streaming, sub-500ms latency. SRT is designed for video transport , contribution feeds, remote production, point-to-point delivery. They serve different use cases.

We often use both: SRT for transport from the venue to our MCR, and WebRTC for final delivery to viewers who need ultra-low latency (interactive Q&A, live auctions, sports betting).

How SRT Replaced Satellite

This is the biggest shift in broadcast technology over the past decade. Satellite contribution , sending a live video feed from a venue to a broadcast centre via satellite uplink , was the standard for decades. It's reliable, but it costs thousands of dollars per hour, requires booking transponder time days in advance, and needs a satellite truck or flyaway kit at the venue.

SRT delivers equivalent reliability over the public internet or bonded cellular at a fraction of the cost. We've covered this transition in detail in our article on SRT replacing satellite transmission in global broadcasting.

The use cases where satellite still makes sense are narrowing every year: truly remote locations with zero cellular coverage, and ultra-high-bandwidth requirements (uncompressed 4K) where internet infrastructure can't keep up. For everything else , and that includes most events in the GCC , SRT over bonded cellular or dedicated internet has replaced satellite entirely.

The MCR Workflow , How We Use SRT in Practice

Here's where SRT fits in our day-to-day production at Creative Broadcast Agency.

Venue to MCR: The Contribution Feed

Cameras at the event venue feed into a hardware encoder (Haivision, LiveU, or Kiloview depending on the setup). The encoder compresses the video in H.265 and transports it via SRT to our Master Control Room in Dubai.

The SRT connection runs over bonded cellular (multiple 5G SIM cards from different UAE carriers combined into a single pipe) or dedicated fibre, depending on venue infrastructure. SRT's error correction handles the variability of both , whether we're on stable fibre at DWTC or bonding cellular signals at an outdoor desert event.

MCR Processing

In our MCR, the SRT feed is decoded, and our production team adds broadcast graphics, lower thirds, sponsor overlays, replay packages, and transitions. This is the critical advantage of the MCR model: production happens in a controlled, permanent facility , not at the mercy of venue conditions.

MCR to Platforms: Distribution

From the MCR, the finished programme feed is distributed to streaming platforms (YouTube, LinkedIn, Facebook, Twitch) and broadcast partners. Each destination receives an output optimised for its specific requirements , bitrate, codec, and delivery protocol.

For broadcast partners who need contribution-quality feeds, we deliver via SRT with individual encryption keys. For platform delivery, we output to each platform's preferred ingest protocol.

Why This Beats Direct-From-Venue Streaming

The traditional approach , encoding at the venue and streaming directly to YouTube or Facebook , has a fundamental problem: you're dependent on the venue's internet for the entire production. If that connection drops, every platform goes down simultaneously.

With SRT to our MCR, the venue connection only needs to support a single SRT uplink (typically 8-15 Mbps for 1080p60 H.265). The MCR handles the multi-platform distribution over its own dedicated internet infrastructure. If the venue connection has a brief interruption, SRT's error correction recovers it. If a streaming platform has an issue, the MCR team re-routes without affecting other outputs.

This workflow reduces dependency on venue internet, ensures consistent quality across all platforms, provides centralised monitoring, and gives us redundancy at every level.

Equipment That Supports SRT

The SRT ecosystem is enormous. Here's what we use in production:

Encoders: Haivision Makito X4, Kiloview P series, Magewell Ultra Encode, LiveU Solo and LU800, Univiso UV100/UV2004. These range from compact field encoders to rack-mount units for permanent MCR installation.

Decoders: Haivision Makito X, Kiloview D350, Univiso UV2004, Magewell Pro Convert. The decoder in our MCR receives the SRT stream and outputs clean baseband video for production.

Bonded Connectivity: LiveU bonding units (combining multiple 5G SIMs), Teltonika RUTX12/TRB500 cellular routers, Peplink MAX multi-WAN routers. These provide the reliable internet pipe that SRT transports over.

Software: vMix, OBS Studio, Wirecast, and most cloud platforms (AWS MediaConnect, Azure Media Services) all support SRT ingest and output natively.

Switching/Production: Blackmagic ATEM, Tricaster, and vMix all accept SRT inputs directly, meaning you can receive remote camera feeds via SRT without any external decoder hardware.

SRT at the Esports World Cup

The Esports World Cup in Riyadh was one of our largest SRT deployments. Here's how the protocol was used across the production:

Multi-arena connectivity. Five competition arenas were connected to the central MCR via fibre and SRT streams. Every game observer feed, player camera, and analysis desk output was transported over SRT to ensure consistent, low-latency delivery to the production switcher.

Remote contribution. We deployed 5G bonding units for roaming reporters throughout the venue complex. These units sent live SRT feeds into the OB truck for real-time integration into the broadcast , interviews, crowd reactions, and behind-the-scenes coverage all transported over SRT.

Global rights distribution. International broadcasters and TV channels that had purchased rights received clean SRT contribution feeds with individual encryption keys. This replaced the satellite distribution model that previous events of this scale would have required.

Live streaming. The finished programme feed was distributed from the MCR to YouTube and online platforms, with SRT used throughout the contribution pipeline to ensure the feed arriving at the MCR was broadcast-grade.

With $27 million in prize money, millions of viewers, and international TV broadcast commitments, there was zero tolerance for stream failures. SRT's resilience delivered , three months of continuous operation without a transport-level failure.

How CBA Uses SRT , Real-World Deployments

As an SRT Alliance member, Creative Broadcast Agency has deployed SRT on hundreds of productions across the GCC. Here are two flagship examples that demonstrate the protocol at scale.

Esports World Cup, Riyadh

The Esports World Cup in Riyadh was a three-month, multi-arena competition with $27 million in prize money and millions of concurrent viewers worldwide. Our SRT deployment included:

• 12+ simultaneous SRT contribution feeds running from five competition arenas into the central production MCR
• SRT latency set to 120 ms on fibre-connected arena feeds, 200 ms on bonded 5G roaming units
• AES-256 encryption on every feed, with individual passkeys per international broadcast rights holder
• 5G bonded SRT links via LiveU LU800 units for roaming reporters and behind-the-scenes coverage , four bonded Zain and STC SIMs per unit, SRT as the transport layer
• Zero transport-level failures across the entire three-month tournament

COP28, Expo City Dubai

The COP28 climate summit at Expo City Dubai required broadcast-grade live coverage of plenary sessions, press conferences, and side events across a sprawling venue complex:

• 8 SRT contribution feeds from multiple stages and press areas into CBA's MCR
• SRT latency at 150 ms over dedicated fibre within Expo City, with bonded 5G backup on every link
• Simultaneous distribution to international news organisations and government channels via SRT with per-recipient encryption keys
• Feeds integrated with broadcast graphics, multilingual lower thirds, and sponsor overlays in the MCR before redistribution

Both deployments demonstrate why SRT has replaced satellite for large-scale event contribution. The protocol handled demanding network conditions , crowded venue Wi-Fi, shared cellular spectrum at capacity events, and long-distance fibre runs , without a single frame drop reaching the production switcher.

For more on our live event streaming and low-latency streaming services, see our service pages.

Getting Started with SRT

If you're evaluating SRT for your production workflow, here are the practical steps:

Test with free tools. OBS Studio supports SRT output natively. SRT Alliance's open-source library is available on GitHub. You can set up an SRT stream between two computers in minutes.

Start with a single contribution feed. Replace one RTMP connection with SRT. Use it for the feed from your venue to your production facility. You'll immediately notice the difference in stream stability.

Configure your latency. Start with 200ms and adjust based on your network conditions. Lower if you need faster delivery, higher if the network is unreliable.

Enable encryption. Always. The performance impact is negligible, and the security benefit is significant. Use AES-256.

Monitor with SRT stats. SRT provides detailed real-time statistics: round-trip time, packet loss, retransmission rate, available bandwidth. Use these to diagnose network issues and optimise your configuration.

The Bottom Line

SRT streaming is the standard for professional live video transport. It replaces satellite for contribution, replaces RTMP for streaming, and provides the security, reliability, and low latency that broadcast production demands.

As an SRT Alliance member, Creative Broadcast Agency has built our entire production infrastructure around this protocol. From our MCR in Dubai to 5G bonded remote production across the GCC, SRT is the transport layer that makes everything work.

If you're planning a live production and want to understand how SRT fits into your workflow, get in touch. We'll show you exactly how it works , and why it matters for your specific use case.

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Internal Links to Include:

• SRT Streaming in 2026: What Broadcasters Need to Know , companion article
• SRT Replacing Satellite Transmission
• Why Low Latency Matters in Esports Broadcasting , SRT gaming article
• Live Event Streaming
• Mobile 5G Streaming
• Low Latency Streaming
• Esports World Cup Case Study

External Links:

• SRT Alliance
• SRT Alliance Members , CBA listed
• SRT Open Source on GitHub