SRT (Secure Reliable Transport) is an open-source transport protocol that moves live audio and video over UDP-based IP connections. It adds packet-loss recovery, jitter handling, a configurable latency buffer and optional AES encryption. SRT is most useful for moving live video between locations across unpredictable networks; it cannot fix insufficient bandwidth, broken connectivity or incompatible endpoint settings.
How SRT streaming works
A source sends media packets to a receiving endpoint over a UDP-based connection. The endpoints also exchange control information about packets and network conditions. If the receiver detects missing packets, it can ask the sender to retransmit them. This process is called Automatic Repeat reQuest (ARQ), and the Haivision SRT project README describes it as SRT’s primary recovery mechanism.
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The receiver’s latency buffer gives retransmitted packets time to arrive before the media is due for playback. That buffer is the central trade-off: allowing more time for recovery can make a stream more resilient to packet loss, but adds transport delay. Recovery also requires network capacity; SRT cannot deliver packets if the path or endpoint cannot support them.
Optional forward error correction
SRT implementations can also support Forward Error Correction (FEC), including ARQ-only, FEC-only and combined FEC-plus-ARQ modes through a packet-filter API and plugin described by the Haivision project. FEC is an implementation capability, not a feature to assume is enabled in every SRT product.
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- 【Multi-protocol and Multi-platform Compatibility】- Fully compatible with streaming protocols such as HTTP, RTSP, RTMP(S), SRT, HLS(M3U8), MP4, Multicast(UDP, RTP, PTL), FLV, WebRTC, TRTC, ICECAST, it can simultaneously output 4 video streams with different protocols and push them to live streaming platforms such as YouTube, Facebook, Twitch, and Vimeo with one click. Simultaneous live streaming across multiple platforms can be achieved without additional equipment.
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Optional encryption
SRT can protect a stream with AES encryption when configured. FFmpeg’s SRT protocol documentation specifies a shared passphrase of 10 to 79 characters when encryption is enabled. Both endpoints need compatible encryption settings and the same secret. Encryption can also use endpoint processing capacity, so check performance as well as connectivity when enabling it.
When to use SRT
Consider SRT for live contribution or distribution links over public or otherwise variable IP networks when packet recovery, jitter handling or encryption matter more than achieving the lowest possible transport delay. Common deployment patterns include sending a contribution feed between sites, aggregating sources, and handing a stream from a gateway to recording, IPTV or digital-signage systems.
SRT may not add enough value to justify its configuration overhead on a stable, protected LAN or WAN. Haivision notes that conventional UDP video can work well on protected networks where packet loss is limited or addressed by FEC. Choose based on the actual path and workflow, not on the assumption that SRT is automatically better for every live stream.
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What SRT does—and does not—provide
SRT is a transport, not a complete streaming service. It does not encode a camera signal, provide a viewer-facing website, distribute adaptive-bitrate playback or replace production software. A working workflow still needs compatible source and destination applications or devices, media encoding and muxing, and downstream distribution if viewers need to watch the stream.
Caller, listener and rendezvous modes
SRT offers three connection modes. They describe how endpoints establish a connection—not which device is the media source.
| Mode | Connection behavior | What to check |
|---|---|---|
| Caller | One endpoint initiates the connection. | Confirm the destination is reachable from the caller and that network rules permit the connection. |
| Listener | One endpoint waits for an incoming connection. | Make the listening destination port accessible under the network’s firewall and NAT rules. |
| Rendezvous | Both endpoints coordinate connection attempts. | Check that both endpoints and the intervening network support the mode; it does not remove all firewall or NAT requirements. |
SRT uses bidirectional UDP. Depending on the topology, deployment can require an accessible destination port, permitted UDP traffic in both directions, and NAT or port-forwarding rules. Rendezvous may suit some symmetric network configurations, but it is not a guarantee that a connection will pass through every firewall.
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How to tune latency and bandwidth
There is no universal SRT latency figure that applies to every workflow. Latency is a configured recovery window, not an end-to-end delay guarantee. The SRT Alliance deployment guide states that, in the configuration it describes, the higher source or destination latency setting is used. Confirm the behavior for the deployed implementation and device.
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Scan for outdated or missing drivers - takes under a minuteDriver Scan →Clear out junk files and repair common Windows errorsFree Scan →- Measure the path. Establish the actual round-trip time, packet loss, media bitrate and observed buffer behavior between the endpoints.
- Set a recovery window. Give retransmissions enough time to arrive, then assess the resulting transport delay. Increasing latency can help when late packets are being missed, but it increases delay.
- Check capacity. Compare the stream’s bitrate and recovery traffic with the path’s available bandwidth. If the path is constrained, lowering video bitrate may help; increasing recovery bandwidth overhead may also help, but requires capacity.
- Change one variable at a time. Observe packet, latency and buffer statistics after each adjustment so you can identify which change affected the result.
- Verify both endpoints. Check that source and destination settings agree where required, including encryption, connection mode and port configuration.
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 a promise that other applications use the same value.
What to monitor during a stream
Use statistics from both endpoints to distinguish a path problem from a configuration problem. Track:
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- Connection and stream state, including repeated reconnections.
- Send and receive bitrate.
- Packet loss, retransmissions and late or skipped packets.
- Round-trip time and configured latency.
- Send and receive buffer behavior, especially buffers that repeatedly exceed the available latency window.
Rising errors, repeated reconnects, excessive skipped frames or buffers that regularly outrun the recovery window call for investigation of path capacity and endpoint settings. The deployment guide identifies increasing latency, lowering video bitrate and increasing bandwidth overhead as possible responses to excessive skipped or dropped frames; test changes against observed statistics rather than applying them blindly.
Independent reader supportYour contribution helps us test, update, and keep practical guides available for everyone.SRT compared with other delivery choices
Compare technologies according to the job they perform. SRT is designed for contribution and inter-system transport; HLS uses segmented HTTP delivery for playback. RTMP, WebRTC, SRT and HLS can occupy different parts of a live workflow, so one should not be treated as a universal replacement for the others.
| Decision factor | Questions to ask |
|---|---|
| Transport delay | How much delay can the contribution or delivery link tolerate? |
| Loss recovery | Does the path need retransmission, FEC, or another recovery approach? |
| Encryption | Can both endpoints configure compatible encryption and manage a shared secret? |
| Network access | Can the required UDP traffic pass the firewall, NAT and port rules? |
| Bandwidth overhead | Is there capacity for the media stream and any recovery traffic? |
| Audience and workflow | Is this a point-to-point contribution link, or does the workflow need broader viewer distribution? |
Troubleshooting common SRT problems
The endpoints do not connect
Check that caller, listener or rendezvous mode is configured as intended, the destination address and port are correct, and the network allows the required bidirectional UDP traffic. Review firewall, NAT, filtering and port-forwarding rules for the actual topology. Rendezvous does not eliminate those checks.
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Video has skipped frames or dropped packets
Inspect loss, retransmission, latency and buffer statistics at both ends. If packets arrive too late for playback, test a larger recovery window. If the path is short on capacity, test a lower video bitrate or a supported increase in recovery bandwidth overhead. Make adjustments in measured steps.
Latency is higher than expected
Review the latency configuration at both endpoints and the implementation’s rules for selecting it; in the deployment guide’s described configuration, the higher source or destination value is used. Then check round-trip time, buffer behavior and any downstream stages, because SRT’s configured recovery window is not a guarantee of total end-to-end delay.
Encryption prevents the connection or stream from working
Confirm encryption is enabled and configured compatibly on both endpoints, and that both use the same passphrase. For FFmpeg’s documented SRT encryption option, the passphrase must be 10 to 79 characters. If the stream runs but endpoint performance suffers, assess processing capacity as well.
Performance changes after enabling FEC
First verify that the specific implementation supports FEC and that it is actually enabled. The Haivision project describes ARQ-only, FEC-only and combined modes; these are not interchangeable assumptions across every SRT product. Compare endpoint statistics and available bandwidth before selecting a mode.
StreamNeo is a different kind of streaming tool
SRT moves a live feed between compatible endpoints. StreamNeo addresses a different job: it keeps uploaded videos or a playlist looping as a 24/7 YouTube live stream from the cloud. It is not an SRT transport, a camera encoder or a way to send an SRT feed to other platforms. If your goal is an always-on YouTube stream from uploaded video, visit StreamNeo or start the first-day-free trial, with no card required, at app.streamneo.com/register.
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