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What Is SRT? A Guide to the Secure Reliable Transport Protocol

SRT is an open-source transport protocol that carries live media and other data while managing packet timing and loss recovery. Learn its latency trade-offs, encryption scope, and optional features.

By DocumentaryTube Team 5 min read
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SRT (Secure Reliable Transport) is an open-source transport protocol that carries live audio and video—and other data—over IP networks. It is designed to manage packet timing and recover from network loss. It is not a video codec, a streaming platform, or a device: it transports media without specifying its format or quality.

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 for reliability and timing. The payload can use different codecs and formats; SRT does not dictate resolution, frame rate, or compression.

In a video workflow, SRT is typically used between sending and receiving endpoints—for example, to contribute a live feed across a network. Both endpoints need compatible SRT support and suitable configuration. The protocol itself is not a hosted streaming service and does not create or encode the video.

How SRT handles packet loss and jitter

SRT’s receiver buffers packets so it can preserve their timing and order while allowing time for recovery. The project says the protocol adapts to changing network conditions and compensates for jitter and bandwidth fluctuation. The buffer helps, but it cannot make a poor connection unlimited: recovery depends on the delay budget, network round-trip time, packet loss, and available bandwidth.

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ARQ: request missing packets again

With automatic repeat request (ARQ), the receiver reports a missing packet and the sender retransmits it. This can recover loss without adding constant redundancy to every packet, but the missing data must arrive within the receiver’s recovery window. Retransmissions also need a usable return path and spare capacity; if the network is congested, they may compete with the live stream.

FEC: send redundancy for reconstruction

Forward error correction (FEC) adds redundant information so a receiver can reconstruct some lost data without waiting for a retransmission. This consumes bandwidth, and recovery delay depends on the packet-group size and stream bitrate. The SRT project documents configurations that combine FEC and ARQ, using redundancy and retransmission as complementary recovery methods.

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Choosing a recovery approach

There is no universally best setting. Consider how much end-to-end delay the application can tolerate, the path’s round-trip time and jitter, the bandwidth available for retransmissions or FEC, the amount of loss, and whether multiple paths are available. The project documentation describes an efficient retransmission algorithm intended to reduce repeated retransmission overhead and an aggressive algorithm that can favor recovery delay when bandwidth can accommodate the overhead. Exact options and behavior can vary by library version.

Latency is a setting, not a guarantee

SRT deliberately trades some delay for a chance to absorb jitter and recover lost packets. A shorter receiver buffer leaves less time for retransmission or FEC reconstruction; a longer one can increase recovery opportunity but adds delay. The right value depends on the application and the network path, so a configured SRT latency is not a promise of end-to-end playback latency.

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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 universal recommendations or measured end-to-end results. The API documentation says the negotiated receiver buffering delay is set during the handshake using the greater of the local receiver latency and the peer’s latency setting.

What SRT encrypts

The SRT protocol draft specifies AES in counter mode (AES-CTR) encryption for data-packet payloads; packet headers are not encrypted. Key material is exchanged during connection setup, with rekeying also specified. The API documentation says incompatible passphrase settings can cause a connection to be rejected by default.

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Payload encryption is useful, but it does not mean every part of every packet is hidden or remove other security responsibilities. Configure compatible encryption settings at both endpoints and consider the protection needed for the surrounding network, applications, and metadata.

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Optional SRT features and where they fit

  • Rendezvous mode: Both endpoints initiate the connection toward one another. The project describes this as a way to traverse certain firewall arrangements without permanently opening externally accessible ports. It still requires compatible endpoint configuration and does not work around every firewall or network policy.
  • Stream ID: An application can use this identifier to distinguish streams arriving at the same IP address and UDP port. It is an application-level way to direct or identify a connection, not a codec or a separate transport.
  • Connection bonding: Multiple IP paths can be used in documented broadcast or main/backup modes. Bonding may add path resilience, but it requires multiple suitable paths and support at the endpoints.

These are deployment choices, not requirements for every SRT connection.

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SRT compared with TCP

The SRT project characterizes SRT as using UDP with reliability improvements and lower latency than TCP/IP in its stated design context. That is a description of the protocol’s intended approach, not a quantified head-to-head performance result. Actual performance depends on configuration, endpoints, and network conditions; no single latency or reliability outcome follows from choosing SRT alone.

Software, tools, and professional endpoints

SRT is an open-source protocol implementation, not a required hardware purchase. Software endpoints can use the project’s library and tools. Dedicated encoder or decoder hardware can also serve as an endpoint in professional contribution or distribution workflows, but it is not a prerequisite established by the project.

The repository includes sample applications such as srt-live-transmit and srt-file-transmit. The project marks these samples as instructional, not production-ready; production deployments should use an implementation and operating setup appropriate to their reliability and security requirements.

Where SRT fits in a YouTube workflow

SRT is a transport protocol between compatible endpoints; it does not itself keep a YouTube channel live from uploaded videos. For a different need—continuous playback of uploaded videos as a YouTube live stream—StreamNeo is a cloud service: upload a recording or create a playlist, add your YouTube stream key, and go live. It runs continuously in the cloud, with one flat price per slot for uploaded quality up to 4K 60fps, and the first day is free with no card. That is a separate workflow from SRT contribution.

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Or let it run in the cloud

If your goal is an always-on YouTube channel playing uploaded videos, StreamNeo lets you upload a video or build a playlist, add your YouTube stream key, and go live. Nothing has to stay on at home; it streams uploaded quality up to 4K 60fps at one price per slot, and it automatically recovers if YouTube drops the stream. The first day is free with no card. Monthly: $9.99 per month.

Start your free day with StreamNeo.

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