SRT (Secure Reliable Transport) is an open-source transport protocol for moving live audio and video over IP networks. It uses UDP and adds packet-loss recovery, jitter handling, a configurable latency buffer, and optional AES encryption. It is most useful for sending live video between locations over an unpredictable network when recovering missing packets matters more than keeping transport delay as low as possible.
SRT cannot repair an inadequate network or replace an encoder, production system, or viewer-facing streaming platform. Its benefit depends on compatible endpoints, sufficient bandwidth, appropriate settings, and a network path that allows the connection.
How SRT streaming works
A source sends media packets to a receiving endpoint over UDP. Alongside the media, SRT endpoints exchange control information about packets and network conditions. The receiver can detect missing packets and ask the sender to retransmit them using Automatic Repeat reQuest (ARQ), SRT’s primary recovery mechanism.
A configured latency buffer gives retransmitted packets time to arrive before the receiver needs to play them. This is the central tradeoff: a longer recovery window can help with late packets, but adds transport delay. SRT does not remove packet loss from the underlying network, and recovery cannot succeed if the path, bandwidth, or endpoint setup is inadequate.
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Optional forward error correction
The Haivision SRT project describes a packet-filter API and a Forward Error Correction (FEC) plugin, with ARQ-only, FEC-only, and combined FEC-plus-ARQ modes. FEC is an implementation capability, not a feature to assume is enabled in every SRT product. Check the documentation for the specific software or device you deploy.
Encryption
SRT can protect a stream with AES encryption when configured. In FFmpeg, the documented setup uses a shared passphrase between endpoints; it must be 10 to 79 characters long. Configure compatible encryption settings and the same secret on both ends. Encryption can also use endpoint processing capacity, so check performance as part of deployment.
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When to use SRT
SRT is a strong option for live contribution or inter-system delivery over public or otherwise variable IP networks, especially when packet recovery, jitter control, and encryption matter more than the lowest possible transport delay. Common workflows include sending a live feed point to point, aggregating sources, or passing a stream through a gateway for recording, IPTV, or digital signage.
On a stable, protected LAN or WAN, simpler transport may be adequate. Haivision notes that conventional UDP video can work well on protected networks when packet loss is limited or addressed with FEC. Choose SRT when its recovery and control features solve a real path problem, not simply because it is newer or assumed to be universally better.
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What SRT does not provide
SRT transports media; it does not encode a camera signal, provide a production interface, host a viewer-facing website, or distribute adaptive-bitrate playback. A complete workflow still needs a compatible source and destination, media encoding or muxing, and downstream distribution if viewers need to watch the stream. SRT and HLS, RTMP, and WebRTC may serve different stages of a workflow, so SRT should not be treated as a blanket replacement for them.
Caller, listener, and rendezvous modes
These modes describe how endpoints establish a connection, not which device is the media source.
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| Mode | Connection behavior | What to check |
|---|---|---|
| Caller | One endpoint initiates the connection. | Confirm that the destination can be reached from the caller’s network. |
| Listener | One endpoint waits for an incoming connection. | Confirm that the listening port is accessible and permitted by firewall and NAT rules. |
| Rendezvous | Both endpoints coordinate connection attempts. | Check support at both ends and verify that the actual firewall and NAT behavior allows the connection. |
SRT uses bidirectional UDP. Depending on topology, deployment may require an accessible destination port, permitted UDP traffic in both directions, and NAT or port-forwarding configuration. Rendezvous can fit some symmetric network arrangements, but it does not automatically bypass firewall work. Check the equipment and network rules in use rather than assuming one mode eliminates them.
Latency, bandwidth, and tuning
SRT latency is a configured recovery window, not a universal end-to-end delay guarantee. The SRT Alliance deployment guide says that, in its described configuration, the higher source or destination latency setting is used. Confirm how the particular implementation handles the settings before deployment.
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FFmpeg documents a recovery-bandwidth-overhead option with a default of 25 percent. That is an FFmpeg option default, not a universal SRT setting or guarantee. Additional recovery capacity may help retransmissions, but it requires available bandwidth. Tune using measured round-trip time, packet loss, bitrate, and observed buffer behavior rather than selecting a value in isolation.
A practical tuning sequence
- Measure the path. Record round-trip time, packet loss, available bandwidth, and the bitrate the source is sending.
- Start with compatible endpoint settings. Confirm connection mode, latency, encryption, and any recovery options at both ends.
- Watch the stream statistics. Review packet loss and retransmissions, late or skipped packets, latency, and send/receive buffer behavior.
- Change one relevant setting at a time. If packets arrive too late, test more latency. If the path is constrained, test a lower video bitrate or, where supported and appropriate, more recovery-bandwidth overhead.
- Recheck the outcome. Compare statistics and stream behavior after each change; more latency and recovery traffic have system-level costs.
What to monitor
Check both endpoints, not just the sender. Useful signals include stream state, bitrate, packet loss and retransmissions, round-trip time, configured and observed latency, and send/receive buffer behavior. Rising errors, repeated reconnections, excessive late or skipped packets, or buffers repeatedly exceeding the available latency window indicate that the path or settings need investigation.
- Loss and retransmissions: Determine whether recovery requests are increasing and whether retransmitted packets arrive in time.
- Latency and buffers: Check whether the recovery window fits the path’s round-trip time and observed packet behavior.
- Bitrate and capacity: Confirm that the path can carry the media plus the recovery traffic the configuration needs.
- Connection state: Look for repeated disconnects that may point to reachability, filtering, or endpoint configuration rather than encoding quality.
Common SRT problems and fixes
| Symptom | Likely area to investigate | Next check |
|---|---|---|
| Endpoints cannot connect | Connection mode, reachability, firewall, NAT, or port configuration. | Verify caller/listener roles, accessible destination port, and permitted bidirectional UDP; check whether forwarding or filtering rules are needed. |
| Frequent disconnects or reconnects | Unstable path or connection setup. | Review endpoint connection settings and network state; inspect stream statistics at both ends. |
| Skipped or late frames | Recovery window too short for the path, packet loss, or insufficient capacity. | Measure round-trip time and loss; test more latency, lower video bitrate, or increased recovery overhead if the implementation supports it. |
| Encryption negotiation fails | Incompatible encryption configuration or mismatched secret. | Confirm encryption is enabled compatibly and the same passphrase is configured at both endpoints; for FFmpeg, verify the documented 10–79 character length. |
| Performance worsens when encryption is enabled | Endpoint processing capacity. | Check CPU or device load at both ends and confirm the selected endpoints can handle the configured media and encryption. |
Choosing SRT in a live workflow
Compare transport options against the job they must do: acceptable delay, packet-loss recovery, encryption, endpoint support, firewall and NAT complexity, bandwidth overhead, and whether the link is a contribution feed or viewer distribution. SRT is designed for moving live media between compatible systems; viewer delivery may require a separate service or protocol. The appropriate choice depends on the whole path rather than a single protocol label.
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