To use SRT for live streaming, send an already encoded stream from an SRT-capable source to an SRT-capable receiver, configure one endpoint as caller and the other as listener (or configure both for rendezvous), match the address, UDP port, latency, and encryption settings, then test the network path before changing video quality. SRT transports live media and can recover from packet loss; it does not capture, encode, or publish the stream by itself.
What SRT does—and what you need before setup
Haivision describes SRT as an open-source transport protocol for live audio, video, and bulk data transfer. It carries a media payload over an IP network and is designed to adapt to changing network conditions and recover from packet loss. It does not require one particular codec or resolution, but the applications on both ends still need to understand the media format being sent.
A typical chain is: camera or other live source → capture device or encoder → SRT connection → receiving application or ingest service. The camera does not send SRT unless it or an attached encoder supports it. SRT is not itself a viewer platform, and connecting two SRT endpoints does not automatically publish a stream to YouTube or another service.
Check these requirements first
- Confirm that both the sending and receiving applications or devices support SRT.
- Check that the receiver accepts the source’s media format and transport stream.
- Identify which endpoint can accept inbound traffic and which can initiate an outbound connection.
- Choose an available UDP port and determine whether host firewall rules or router/NAT forwarding are needed.
- Agree on latency and, if encryption is enabled, the encryption settings and passphrase.
Choose caller, listener, or rendezvous
These modes describe how the SRT connection is established—not which endpoint sends the video. The media sender can be either caller or listener, provided the other endpoint takes the compatible role.
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| Mode | How it works | Network consideration |
|---|---|---|
| Caller | Initiates a connection to a listener’s address and port. | Useful when the caller can make outbound connections but is not directly reachable for inbound traffic. |
| Listener | Binds to a local address and UDP port and waits for a caller. | If behind a firewall or NAT, allow the port and, where needed, forward it to the listener. The remote caller must target the reachable address and port. |
| Rendezvous | Both endpoints attempt to establish the connection. | May suit some firewall-traversal arrangements, but both applications and network paths must be compatible; it will not pass every firewall or NAT. |
For a common public-Internet arrangement, the receiving side is often the listener because it is straightforward to set up a reachable port there. That is a practical convention, not a requirement. Choose roles based on which endpoint can be reached and what the applications support.
Configure an SRT stream step by step
- Verify the media source independently. Confirm the camera, capture device, or encoder produces a working live stream before adding SRT. The SRT application guide recommends isolating source problems first.
- Set complementary connection modes. Configure one endpoint as caller and the other as listener, or use rendezvous at both ends if the applications and networks support it. Do not assume the video sender must be the caller.
- Enter addresses and ports on the right endpoint. A listener binds to its local interface and UDP port. A caller enters the listener’s reachable address and port. With rendezvous, both sides’ addresses and ports must match the applications’ requirements; consult their documentation for adapter or bind-address options.
- Make the listener reachable. Permit the chosen UDP port through the listener’s host firewall. If a router performs NAT, configure the appropriate UDP port forward to the listener. Ensure the address used by the caller is the one actually reachable from its network.
- Choose a starting latency from measured RTT. Measure round-trip time between endpoints, then begin around three to four times that RTT, following the SRT project’s testing guidance. Treat this as a starting point, not a universal optimum. Check live connection statistics and adjust if RTT varies or recovery is insufficient.
- Match encryption settings. If encryption is on, set compatible passphrase and key options at both ends. An enforced-encryption configuration can reject a connection when passphrases differ. Keep the passphrase private and treat it like a credential.
- Start the receiver and sender, then inspect connection statistics. Check whether the connection establishes and monitor RTT and other available SRT statistics while the stream is running, especially over wireless or mobile networks.
Set latency and understand packet recovery
SRT’s receiver latency is a buffer interval: it holds arriving packets long enough for delayed packets to arrive and for missing packets to be requested again before playout. A smaller interval reduces transport buffering but leaves less time for recovery; a larger interval adds delay while allowing more room to absorb jitter and retransmissions. The right balance depends on measured RTT, jitter, loss, and how interactive the production needs to be.
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The SRT API documentation lists a 120 millisecond receiver-latency default for its live transmission configuration. That is a software configuration default—not a measured average, minimum, end-to-end latency claim, or recommendation for every setup. Capture, encoding, decoding, and display add delay outside SRT. Do not treat the protocol’s sub-second positioning as a guarantee of sub-second end-to-end delivery.
Resilience options
- ARQ retransmission: SRT requests that missing packets be sent again. The latency buffer must leave enough time for recovery.
- FEC: Adds recovery data, which may help in suitable network conditions but has bandwidth implications.
- Connection bonding: Uses multiple IP paths. Broadcast bonding sends data redundantly over member links; Main/Backup keeps backup paths standing by.
These are optional resilience techniques, not substitutes for enough bandwidth, compatible configuration, or a stable network path. Bonding requires appropriate multiple paths, and redundant transmission can consume additional capacity.
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Encryption, firewalls, and reachability
SRT supports AES encryption of the stream payload. Both endpoints must be configured compatibly; where encryption is enforced, a passphrase mismatch can block the connection. Encryption does not replace firewall policy or careful credential handling.
For Internet connections, verify the complete UDP route: the listener’s local bind address and port, host firewall, router/NAT forwarding, and the public address and port entered by the caller. Rendezvous can help in some traversal configurations, but actual firewall and NAT behavior determines whether a path works.
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Troubleshoot connection and stream problems
| Symptom | Likely cause | What to check |
|---|---|---|
| Connection never establishes | Incompatible modes, wrong address or port, or unreachable listener. | Confirm caller/listener pairing or compatible rendezvous settings. Verify the listener bind, caller target, UDP firewall rule, and any NAT port forward. |
| Connection works on a local network but not over the Internet | The listener’s UDP port is not reachable through the router or firewall, or the caller is using the wrong address. | Check the public address, host firewall, router forwarding destination, and selected UDP port. |
| Connection fails after enabling encryption | Passphrase or key settings differ, or enforced encryption rejects the mismatch. | Compare both endpoint configurations securely and confirm encryption options are supported and aligned. |
| Video does not appear despite a connection | The source is not producing usable media, or the receiver does not accept the payload format. | Test the source separately, then verify media compatibility through the full chain. |
| Playback stutters or recovery is poor | Latency may be too low for RTT variation or packet recovery; available bandwidth or link quality may be inadequate. | Inspect RTT and connection statistics, test a higher latency based on measured behavior, and check network capacity before altering codec or bitrate settings. |
| Performance changes on wireless or mobile links | RTT can vary on wireless LAN, line-of-sight radio, LTE/4G, and 5G paths. | Monitor RTT over time and tune against observed variation rather than a single measurement. |
Choosing an SRT-capable encoder or capture setup
A software workflow may already include capture and encoding. A hardware workflow might use a frame grabber or capture device feeding a live encoder, or a network encoder that supports the source format and SRT. A generic capture card is not automatically an SRT encoder.
Before buying hardware, verify in the manufacturer’s documentation that it explicitly supports SRT and the intended receiver workflow. Also check supported codecs and resolutions, network interfaces, caller/listener/rendezvous options, and encryption controls. Availability and compatibility depend on the particular model and destination; confirm them rather than inferring SRT support from a product’s general streaming label.
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