To use SRT for live streaming, send an already encoded live stream from an SRT-capable encoder or application to a compatible receiver. Choose which endpoint will call and which will listen, configure the address and UDP port, match encryption settings, and set receiver latency based on measured network round-trip time (RTT). SRT transports and helps recover media packets; it does not capture video, encode it, or publish it to a viewing platform.
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 media between networked endpoints and is content agnostic: it does not require one particular codec or resolution. The SRT project describes its purpose as adapting to changing network conditions and recovering from packet loss, primarily through ARQ retransmission. It also supports FEC and connection bonding. These features can improve resilience, but they cannot compensate for inadequate bandwidth, persistent loss, incompatible endpoints, or a misconfigured receiver. SRT project documentation
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Think of an SRT workflow as three parts: a source and encoder that produce the live media, SRT as the network transport, and a destination that can receive and use the media. A source might be a camera and capture device feeding an encoder, a network encoder, or a software application producing a live stream. Confirm that the media format emitted by the source is supported by the receiving application; SRT itself does not make incompatible media formats compatible.
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- A compatible SRT receiver, decoder, production system, or ingest service.
- Network access between the endpoints, with an appropriate UDP path.
- Matching connection-mode and encryption settings at both ends.
- A way to confirm the source works independently and to inspect SRT connection statistics.
Configure an SRT connection step by step
- Verify both endpoints. Confirm SRT support on the sender and receiver, and check that the receiver can handle the media format and transport stream the sender will provide.
- Choose the connection mode. Decide which endpoint initiates the handshake. Caller mode initiates toward a listener; rendezvous has both sides attempt to connect. The media sender does not have to be the caller.
- Set the address and UDP port. A listener binds a local address and port. A caller targets the listener’s reachable address and port. For rendezvous, both endpoints must have compatible addresses, ports, and any required adapter or bind settings.
- Make the network path reachable. Where a listener is behind a firewall or NAT, allow the chosen UDP port through the host firewall and configure router forwarding to the listener if needed. Ensure the local bind address and externally reachable address/port point to the intended endpoint.
- Set receiver latency from measured RTT. Use the SRT project’s guidance of roughly three to four times measured RTT as a starting point, then monitor live behavior and retune. This is a recommendation to test, not a guaranteed optimum.
- Match encryption settings. If using encryption, configure compatible passphrase and key settings on both endpoints. Keep the passphrase private as you would any credential.
- Test the source first, then start the SRT route. Confirm video and audio work before sending through the transport. Once connected, watch RTT and connection statistics, especially on wireless or mobile links where network conditions can vary.
The SRT application guide recommends verifying the source separately before routing it through srt-live-transmit. Exact labels and controls depend on the encoder, receiver, and software version; use each endpoint’s documentation for the corresponding address, port, bind, latency, and encryption fields. SRT application and project documentation
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Caller, listener, or rendezvous?
These modes describe how the SRT connection is established—not which endpoint sends the video. A media sender can be caller or listener, as long as the other endpoint uses a compatible role.
| Mode | How it connects | When it can fit | Network consideration |
|---|---|---|---|
| Caller | One endpoint initiates a connection to a listener. | Useful when the calling endpoint can make outbound connections but is not directly reachable for inbound connections. | The caller must target the listener’s correct reachable address and UDP port. |
| Listener | The endpoint binds an address and UDP port, then waits for a caller. | Often used at the receiving end in common setups, though it is not a universal requirement. | If behind a firewall or NAT, the listener’s UDP port may need to be allowed or forwarded. |
| Rendezvous | Both endpoints attempt to establish the connection. | Can help with some firewall-traversal setups when both peers and paths are configured compatibly. | It does not guarantee passage through every firewall or NAT; address, port, and adapter settings must align. |
Choose latency without guessing
Receiver latency is the interval packets are buffered before delivery to the receiving application. That time gives delayed packets a chance to arrive and gives retransmission a chance to recover missing packets before playout. Lower latency means less delay but also less recovery time; higher latency adds delay while allowing more room to absorb jitter and request retransmissions. The right balance depends on RTT, jitter, packet loss, and how much interactive responsiveness matters.
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The SRT project’s srt-live-transmit testing guidance suggests latency roughly three to four times measured RTT, followed by monitoring and adjustment if RTT varies. Treat this as a starting recommendation, not a formula that guarantees good results. Wireless LAN, line-of-sight radio, LTE/4G, and 5G connections are examples of paths where RTT can vary. SRT testing and tuning guidance
The SRT API documentation describes a 120 millisecond receiver-latency default for its live-transmission configuration. That is a software default—not an average, minimum, end-to-end latency claim, or universal recommendation. Capture, encoding, decoding, and display also add delay outside the transport. SRT API documentation
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What ARQ, FEC, and bonding contribute
- ARQ retransmission: The receiver can request that missing packets be sent again. This depends on there being enough time in the latency buffer and sufficient network capacity for recovery traffic.
- Forward error correction (FEC): Adds recovery data so losses can be handled without relying only on retransmission. It is an optional resilience method, not a fix for every poor link.
- Connection bonding: Uses multiple IP paths. In Broadcast bonding mode, data is sent redundantly on member links; in Main/Backup mode, backup paths stand by. These modes have different bandwidth and network requirements.
These are options for suitable deployments, not prerequisites for a basic SRT connection. Increasing resilience can involve trade-offs in bandwidth, delay, or network complexity; choose based on the paths available and the behavior you observe.
Encryption, firewalls, and reachability
SRT supports AES encryption of the stream payload. Both endpoints must be configured compatibly. Haivision’s application documentation describes an enforced-encryption option that rejects connections when peers use different passphrases. Protect the passphrase, and do not treat encryption or rendezvous as a replacement for sensible firewall policy.
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When connecting over the public Internet, establish reachability before changing video settings. The SRT project’s guidance calls for the listener to be reachable on its chosen UDP port and recommends appropriate UDP port forwarding on routers using NAT. Check that the listener is bound to the intended local interface, that host firewall rules allow the traffic, and that any router forwarding targets the correct device and port. SRT network configuration guidance
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Troubleshoot an SRT stream
| Symptom | Likely checks and fixes |
|---|---|
| No connection or handshake | Confirm caller, listener, or rendezvous roles are compatible; verify the caller’s target address and port; check the listener’s local bind; allow the UDP port through host firewalls and forward it through NAT where required. |
| Connection works only on the local network | Check the public-facing address and router UDP forwarding, then verify that forwarding reaches the listener’s device and local port. Confirm firewall rules on both the router and host. |
| Connection fails with encryption enabled | Compare encryption enablement and passphrase/key settings at both endpoints. If enforced encryption is configured, a passphrase mismatch can cause rejection. |
| Video or audio does not appear after connection | Test the source on its own, then verify the sender’s media format is accepted by the receiving application. SRT transports the payload but does not convert formats or confirm that the downstream application can decode them. |
| Playback stutters or drops packets | Inspect RTT and connection statistics; check for varying RTT, jitter, packet loss, or insufficient available bandwidth. Test a receiver latency based on measured RTT, then monitor again. More buffering can improve recovery time but increases delay. |
| Latency is higher than expected | Check the configured receiver buffer and account for capture, encoding, decoding, and display delays, which are outside SRT’s transport buffer. |
Do not start by raising bitrate when the connection cannot be established: codec and picture-quality settings do not resolve a blocked UDP path or mismatched connection roles.
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Choosing software or an SRT-capable encoder
A software workflow may already include capture and encoding functions. A hardware workflow may use a 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 the manufacturer’s documentation for explicit SRT support, supported codecs and resolutions, network interfaces, caller/listener/rendezvous options, encryption controls, and compatibility with the intended receiver or ingest endpoint. Product-specific model availability and pricing are not established here, so check current manufacturer listings for those details.
Or let it run in the cloud
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