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SRT stands for Secure Reliable Transport. It is an open-source protocol for sending live audio and video over IP networks, including the public internet. It helps recover missing packets by buffering data and requesting retransmissions, but it does not guarantee uninterrupted delivery or a particular end-to-end delay.
What SRT is—and what it is not
SRT is a transport protocol: it moves media data between endpoints. It does not define the video codec or file format, so it can carry different kinds of encoded content without replacing the encoder or decoder. SRT is open source and is used in live contribution, remote production, distribution, and cloud workflows. The Haivision SRT project also describes file-based media and generic bulk data transfer.
Despite the word “Secure” in its name, the name alone does not tell you whether a particular connection is encrypted. SRT supports AES encryption, but encryption must be configured and supported by the endpoints.
How SRT works
- The sender transmits media packets. An SRT sender sends packets to a receiving endpoint over an IP connection. SRT wraps the content for transport; the media’s codec is handled separately.
- Buffers account for network variation. Packets can be delayed or arrive out of order. Buffers at the endpoints hold data so the receiver can restore the sequence and deliver it to the decoder at the intended timing.
- The receiver requests missing packets. Using Automatic Repeat reQuest (ARQ), the receiver detects a gap and asks the sender to retransmit the missing packet.
- The receiver delivers recovered data in time—or proceeds without it. A retransmission can help only if it arrives before the configured latency deadline and the connection has enough spare capacity to carry it. Otherwise, the packet cannot be recovered in time for that playback.
Retransmission is therefore a trade-off, not a way to make a weak connection unlimited. It consumes bandwidth, and a larger buffer gives packets more time to arrive at the cost of added delay.
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Latency: choosing a recovery window
Haivision’s SRT 1.5.4 latency documentation defines a configurable latency value from 20 to 8,000 milliseconds. It says the higher latency value configured at either endpoint is used. These are version-specific figures and vendor guidance, not a universal performance guarantee.
For a fairly good network with 0.1–0.2% packet loss and no significant burst loss, Haivision gives four times the round-trip time (RTT) as a rule of thumb for setting latency. That guidance is conditional: burst loss, fluctuating bandwidth, and the actual workflow can make an RTT-based estimate insufficient.
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- More latency: gives retransmissions a longer window to arrive, but increases the delay in the media path.
- Less latency: reduces buffering delay, but leaves less time to recover delayed or missing packets.
- More loss or burstier loss: may require a different configuration; retransmissions still need time and bandwidth headroom.
Use the endpoint documentation as a starting point, then monitor and test the actual network and workflow. A single latency value cannot account for every link condition.
Where SRT is used
SRT is suited to workflows that need to move live media across networks where delay, packet loss, or changing bandwidth may affect delivery. Official materials describe broadcast contribution, remote production, distribution, and cloud workflows. It can be implemented in software or supported hardware; a dedicated encoder is not necessary just to understand or try the protocol. Haivision’s technical implementation guide lists the Makito X4 among related products, while the SRT project itself is available as open-source software.
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SRT compared with RTMP or RIST
No transport is a universal winner. To assess SRT alongside RTMP or RIST for a specific workflow, compare how each handles packet loss, the latency budget, bandwidth overhead, encryption configuration, endpoint compatibility, and any available multipath redundancy. Haivision’s project materials describe ARQ, FEC, and connection bonding for SRT, but those capabilities alone do not establish how it will perform against another protocol on a particular network.
The SRT Alliance FAQ discusses comparisons with RTMP and RIST, but Alliance descriptions should be distinguished from neutral, side-by-side performance evidence. No independent cross-protocol benchmark is established here; test compatible endpoints under conditions representative of your own link before choosing.
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Common misconceptions
- “SRT is a video codec.” It is a transport protocol; it does not define the media codec.
- “SRT guarantees a stable stream.” Recovery depends on packets arriving within the latency window and on sufficient bandwidth.
- “Every SRT connection is encrypted.” AES encryption is supported, but the connection’s configuration matters.
- “Lower latency is always better.” A shorter buffer leaves less time for retransmission; the useful setting depends on the link and workflow.
How SRT relates to keeping a YouTube channel live
SRT is a transport option for compatible endpoints; it is not, by itself, a managed service for continuously looping prerecorded videos to YouTube. If your goal is a 24/7 YouTube stream from uploaded recordings rather than an SRT production workflow, StreamNeo is a separate cloud service: upload a video or build a playlist, add your YouTube stream key, and go live. It streams uploaded videos to YouTube, not from a camera.
Or let it run in the cloud
With StreamNeo, nothing has to stay on at home. Each slot streams your upload as made, up to 4K 60fps, at one flat price per slot; it can automatically recover if YouTube drops the stream. The first day is free with no card. Monthly pricing is $9.99 per month.
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