SRT (Secure Reliable Transport) is an open-source transport protocol that carries live video, audio, or other data across IP networks while helping manage packet loss, timing, and optional payload encryption. It is not a video codec, a streaming platform, or a device. In a streaming workflow, it moves an encoded media stream between endpoints; the codec, resolution, and frame rate are determined elsewhere.
What SRT does—and what it does not
The Haivision SRT project describes SRT as a protocol for low-latency live video and audio streaming and generic bulk data transfer. It runs over UDP and adds mechanisms intended to help media reach the receiver in order and on time despite jitter, packet loss, or bandwidth fluctuation. That design does not guarantee a particular end-to-end delay: the result depends on the endpoints, configuration, and network path.
- It is a transport protocol: it carries a payload between applications or devices.
- It is payload-agnostic: it does not prescribe a codec, container, resolution, or frame rate.
- It is not a streaming service: SRT alone does not provide a destination channel, audience, or video player.
- It is not inherently a hardware purchase: the open-source project provides a library and sample tools; dedicated encoder or decoder hardware may be used as an endpoint in some professional workflows, but is not a protocol prerequisite.
The project’s README calls SRT a protocol for “ultra low (sub-second) latency” live media. Treat that as the project’s description of its design goal, not a promise that every SRT workflow will deliver sub-second playback.
How SRT handles packet loss and timing
SRT’s receiver uses a latency buffer. By holding packets briefly, it can preserve their timing and sequence while allowing later packets to arrive—or giving the sender time to retransmit missing ones. This buffer is the central trade-off: more time can improve the chance of recovering loss before playback, while less time leaves less room for recovery.
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ARQ: request missing packets
With automatic repeat request (ARQ), the receiver identifies missing data and requests retransmission from the sender. ARQ needs a usable return path, enough bandwidth to carry the retransmitted traffic, and enough time remaining in the configured recovery window for the packet to arrive before it is needed. SRT documentation describes efficient and aggressive retransmission algorithms; the latter can favor recovery delay at the cost of additional bandwidth overhead. Exact options and behavior can vary by library version.
FEC: add redundancy in advance
Forward error correction (FEC) adds redundant information so a receiver can reconstruct some lost data without waiting for a retransmission. The recovery depends on receiving enough information from the relevant group. FEC therefore consumes bandwidth and can add delay related to group size and stream bitrate. SRT documentation also describes combined FEC and ARQ operation.
| Method | How recovery works | Main trade-off |
|---|---|---|
| ARQ | Receiver reports a missing packet; sender retransmits it. | Requires a return path, spare retransmission capacity, and enough recovery time. |
| FEC | Redundant data can let the receiver reconstruct losses. | Uses extra bandwidth and may add group-related delay. |
| Combined FEC and ARQ | Uses both redundancy and retransmission mechanisms. | Can address different loss cases, but still has bandwidth and latency costs. |
Neither ARQ nor FEC is universally better. The choice depends on the latency budget, round-trip time and jitter, available bandwidth, acceptable loss, and whether a return path or multiple paths are available.
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Latency: what the setting means
SRT latency is a recovery and playback buffer setting, not a universal end-to-end performance figure. The receiver intentionally holds packets so they can be played in sequence and missing data may arrive in time. Increasing the recovery window can help on a variable or lossy path; reducing it may suit a tighter delay target if the path can reliably support it.
The Haivision SRT API reference, accessed in 2026, lists receiver-latency defaults of 120 ms in Live mode and 0 ms in File mode. These are library defaults, not recommended values for every stream and not measured playback outcomes. The reference says the negotiated receiver buffering delay is determined during the handshake using the greater of the local receiver-latency setting and the peer latency setting. Check the documentation for the specific library version and mode in use before relying on defaults.
Encryption and security scope
The SRT protocol draft specifies AES in counter mode (AES-CTR) encryption for data-packet payloads. Packet headers are not encrypted. The draft describes exchanging stream-encryption key material during connection setup and rekeying; the API documentation also describes rejecting connections when passphrase settings are incompatible by default.
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Payload encryption is useful, but it does not mean every piece of packet metadata is secret or remove the need to secure endpoints, credentials, and network configuration. Follow the implementation’s current security guidance and configure compatible passphrase settings on both sides.
Optional SRT capabilities
These capabilities are available only where the relevant software or devices support and configure them; they are not requirements for every SRT connection.
Rendezvous mode
Rendezvous lets both sides initiate a connection toward one another. The project describes it as useful for traversing certain firewall arrangements without permanently opened external ports. It still depends on compatible endpoint configuration and network conditions; it is not a general guarantee that every firewall can be traversed.
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Stream ID
A Stream ID lets an application distinguish streams arriving at the same IP address and UDP port. It is an application-level way to identify a stream, not a change to the media codec.
Connection bonding
Bonding uses multiple IP paths. The project documents broadcast and main/backup modes: broadcast can use paths for redundancy, while main/backup provides a standby path. Bonding requires multiple usable paths and support at both relevant endpoints; it is not a benefit available from a single network connection alone.
Independent reader supportYour contribution helps us test, update, and keep practical guides available for everyone.Where SRT fits in a video workflow
SRT can serve as a contribution or distribution transport between compatible software or hardware endpoints, including over networks whose conditions vary. The project also lists generic bulk data transfer. The repository includes examples such as srt-live-transmit and srt-file-transmit, but identifies its sample applications as instructional rather than production-ready. Production workflows should use an implementation and operational setup appropriate to their reliability, security, and support requirements.
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In a gaming or creator workflow, SRT is relevant when a compatible encoder, relay, or receiving service uses it to carry media. It does not by itself capture gameplay, encode a video, create a YouTube broadcast, or guarantee compatibility with a particular platform; verify that every endpoint in the intended path supports the protocol.
SRT compared with TCP
The SRT project characterizes its UDP-based transport with reliability improvements as lower-latency than TCP/IP in its stated design context. That is a project-level description, not a quantified head-to-head benchmark or guarantee for a specific route. TCP and SRT have different transport behavior; which is appropriate depends on the application’s tolerance for delay and loss, the available network paths, and the capabilities of the endpoints.
Common misconceptions
- “SRT is a codec.” It transports a payload; it does not encode or decode the audio or video format.
- “SRT always means sub-second playback.” The project uses sub-second language to describe its design positioning. Configuration, buffering, network conditions, and the rest of the workflow affect actual delay.
- “Encryption hides the entire connection.” The protocol draft specifies payload encryption, while packet headers remain unencrypted.
- “FEC or retransmission makes packet loss irrelevant.” Both need suitable network capacity and configuration; ARQ also needs recovery time and a return path, while FEC adds redundancy.
- “SRT requires a dedicated appliance.” Software implementations exist. Hardware can be an endpoint choice, not a universal requirement.
When to consider SRT
- Consider it when both endpoints support SRT and you need a transport designed to cope with packet loss, jitter, or changing network conditions.
- Evaluate the end-to-end delay target alongside round-trip time and jitter; do not choose a buffer value solely because it is a published default.
- Confirm there is enough capacity for retransmissions or FEC overhead, as applicable.
- Use bonding only when multiple network paths and compatible endpoint support are actually available.
- For a production deployment, verify the implementation version, supported options, security settings, and operational support rather than relying on instructional sample apps.
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