SRT (Secure Reliable Transport) is an open-source protocol that carries live audio and video over IP networks. It is designed to handle packet loss, jitter and changing bandwidth by buffering packets and requesting retransmission of missing data—provided the data can arrive before the configured recovery deadline. SRT is a transport protocol, not a video codec, and it cannot guarantee a trouble-free stream on a poor connection.
What SRT streaming means
SRT moves media between a sending endpoint and a receiving endpoint across an IP connection. It wraps the audio and video payload; it does not define the video codec or media format. SRT is open source and is used in broadcast contribution, remote production and cloud workflows. The project also describes file-based media and general data transfer as possible uses.
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Using SRT does not require a dedicated encoder just to understand or try the protocol: software implementations and supported hardware endpoints are both possible. Haivision lists its Makito X4 among products related to its technical implementation guide, but the protocol itself is not a particular device.
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How SRT works
- The sender transmits media packets. SRT carries the payload over an IP connection without prescribing its codec.
- Buffers absorb timing variation. The receiver holds packets that arrive late or out of sequence, then reorders them for delivery to the decoder at the intended timing.
- The receiver requests missing packets. Using Automatic Repeat reQuest (ARQ), it detects a gap and asks the sender to retransmit the missing data.
- The receiver delivers what arrives in time. Retransmission can repair loss only if there is enough time before the configured latency deadline and enough network capacity to carry the extra packets.
This recovery mechanism trades delay and bandwidth headroom for a better chance of delivering packets despite loss. If the link is congested, losses are bursty, or retransmissions arrive too late, SRT cannot recover every packet.
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Latency: the key setup tradeoff
Haivision’s SRT 1.5.4 documentation gives a configurable latency range of 20–8000 ms. It says to base the setting on the link in use and notes that the higher latency value configured at either endpoint is used. These are version-specific vendor instructions, 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. That is a starting point for the stated conditions, not a setting to apply blindly: burst loss or changing bandwidth can make RTT alone insufficient.
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- CLOSED CAPTIONS, TIMECODE & REST API – Supports embedding CEA-608 and CEA-708 closed captions in live RTMP streams, source timecode over RTMP and SRT, and offers a REST API over Ethernet for external HTTP control — ideal for broadcast automation and accessibility-compliant workflows.
- More latency: Gives retransmissions a longer window to arrive, but adds delay to the workflow.
- Less latency: Reduces the buffer delay, but leaves less time to repair late or missing packets.
- More retransmission traffic: Uses network capacity that may already be limited on a troubled link.
In practice, monitor the link and test settings in the actual sending and receiving workflow. A single fixed value cannot account for every route, loss pattern or bandwidth fluctuation.
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Encryption and security
SRT supports AES encryption of stream payloads. That is a protocol capability, not proof that a given connection is encrypted: the endpoints must be configured accordingly. Confirm encryption settings at both ends when confidentiality is a requirement.
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Where SRT fits
Official SRT materials describe uses including live broadcast contribution, remote production and cloud workflows. The open-source project also describes file-based media and generic bulk data transfer. Whether SRT fits a particular job depends on the endpoints, network conditions and the workflow’s acceptable delay.
The SRT Alliance homepage describes more than 700 organizations as relying on SRT. That figure is the Alliance’s own adoption claim, not an independent market count.
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How SRT compares with RTMP or RIST
There is no universal winner established by the sources cited here. A useful comparison is specific to the workflow, not just a claim that one protocol is “better.” Assess:
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- The loss pattern on the link, including whether loss arrives in bursts.
- The latency budget and the time available to recover packets.
- Bandwidth overhead and available capacity.
- Whether encryption is supported and enabled in the actual configuration.
- Compatibility with the sender, receiver and other workflow equipment.
- Whether multipath redundancy is available and needed.
Haivision’s project materials document ARQ, FEC and connection bonding for SRT. Those features do not establish a neutral performance ranking against RTMP or RIST; the reviewed materials do not provide independent side-by-side benchmark results.
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Keeping a YouTube channel live around the clock
SRT is a transport protocol for moving media between endpoints; by itself, it is not a service for continuously looping uploaded videos to YouTube. If your goal is a 24/7 YouTube channel built from prerecorded videos, StreamNeo is a separate cloud option: upload a recording or build a playlist, add your YouTube stream key once, and go live. It loops the uploaded content from the cloud, so your computer does not need to stay on.
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