To use SRT for live streaming, send an already encoded live feed from an SRT-capable source to an SRT-capable receiver, choose which endpoint initiates the connection, configure matching addresses, UDP ports and encryption, then tune receiver latency against measured network conditions. SRT transports the stream; it does not capture or encode video or publish it to a viewer platform.
What SRT does—and what you need around it
SRT (Secure Reliable Transport) is an open-source protocol for carrying live audio and video over IP networks. Haivision describes it as a transport for live video, audio and bulk data transfer. Its job is to adapt to changing network conditions and recover from some packet loss, primarily through ARQ retransmission, in which the receiver requests missing packets again. The SRT project also supports forward error correction (FEC) and connection bonding. These capabilities can improve resilience, but they cannot overcome insufficient bandwidth, severe loss or incompatible endpoint settings.
SRT is content agnostic: it does not require one particular codec or resolution. However, the applications at both ends still need to agree on a media format they can handle. The chain is typically a camera or other source, a capture device if needed, an encoder, the SRT transport link, and a receiving decoder, production system or ingest service. SRT carries the encoded feed between applications or devices; it is not the camera, encoder or destination platform.
Before configuring the link
- Identify the source/encoder and receiving endpoint, and confirm that both explicitly support SRT.
- Check that the source’s media format and transport are accepted by the receiver.
- Decide which endpoint can make an outbound connection and, if using a listener, who can make its UDP port reachable.
- Agree on latency and encryption settings before testing.
Configure an SRT connection step by step
- Verify the media path independently. Confirm the camera, capture device or other source produces a working feed and that the encoder can output it. The SRT application guide recommends validating the source before routing it through
srt-live-transmit. - Choose a connection mode. Use caller, listener or rendezvous according to which endpoint can initiate traffic and what the network permits. These roles describe the connection handshake, not which endpoint sends the video.
- Enter the address and UDP port on the appropriate endpoint. A listener binds a local address and port; a caller targets the listener’s remote address and port. In rendezvous mode, both endpoints’ addresses and ports must align, and some applications require adapter or bind settings.
- Set receiver latency. Start from measured round-trip time (RTT), then monitor live statistics and adjust for jitter and variation. The SRT project’s testing guidance suggests latency around three to four times measured RTT as a starting point, not a universal optimum.
- Match encryption settings. If encryption is enabled, configure compatible passphrase and key settings at both ends. Treat the passphrase as a credential and protect it accordingly.
- Test the connection before tuning picture quality. Confirm that the endpoints connect and that the receiver gets the expected media. For an Internet link, verify the listener’s UDP port is reachable through host firewalls and any router or NAT. Watch RTT and connection statistics during the test.
Choose caller, listener, or rendezvous
| Mode | How it connects | Network considerations |
|---|---|---|
| Caller | One endpoint initiates a connection to the remote listener. | Useful when the caller can make outbound connections but is not directly reachable for inbound connections. Configure the caller to target the listener’s reachable address and port. |
| Listener | An endpoint binds to a local address and port, then waits for a caller. | The listener must be reachable. A firewall rule and, behind NAT, router UDP port forwarding may be required. The SRT project notes this is commonly used at the receiving end because firewall/router setup can be easier there; it is not a universal rule. |
| Rendezvous | Both endpoints attempt to establish the connection at the same time. | Both peers and their network paths must be configured compatibly. It can help with some firewall traversal setups but does not guarantee passage through every firewall or NAT. |
Do not assume the caller is the video sender. The media sender can be either caller or listener, provided the other endpoint uses the complementary connection role.
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Set latency and understand packet recovery
Receiver latency is the interval SRT buffers incoming packets before passing them to the application. The buffer gives delayed packets time to arrive and allows retransmission requests to recover missing packets before playback. A smaller interval reduces transport delay but leaves less time for recovery; a larger interval adds delay. Choose a balance based on RTT, jitter, packet loss and how interactive the production needs to be.
The SRT project’s testing notes recommend starting with latency roughly three to four times measured RTT, then monitoring whether RTT changes and retuning. This is project guidance to test, not a guarantee. Wireless LAN, line-of-sight radio, LTE/4G and 5G paths are among those specifically noted as having variable RTT.
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The Haivision SRT API documentation describes a 120 millisecond receiver-latency default for its live transmission configuration; the reviewed page does not state a publication year. That figure is a software default—not an average, minimum, end-to-end latency claim or setting suitable for every link. Capture, encoding, decoding and display also add delay outside the SRT transport.
Optional resilience features
- ARQ: requests retransmission of missing packets. Recovery depends on the packets arriving in time and the network having enough capacity.
- FEC: adds recovery data. It can help with loss but has bandwidth requirements.
- Connection bonding: combines 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 path and bandwidth needs.
These are optional tools for suitable deployments, not substitutes for a working source, adequate bandwidth or correct endpoint configuration. SRT’s “sub-second” positioning should not be read as a promise about a particular workflow’s end-to-end delay.
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Encryption and network reachability
SRT supports AES encryption of the stream payload. Both endpoints must be configured compatibly. Haivision application documentation describes an enforced-encryption option that rejects connections if peers use different passphrases. Encryption does not replace firewall policy or safe credential handling.
For an Internet connection where one endpoint listens, make the listener’s local bind address and port, public address and port, host firewall rule, and router/NAT forwarding agree. Rendezvous can assist with some traversal configurations, but the actual firewall, NAT and endpoint behavior still determine whether a path works.
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Troubleshoot in the order that isolates the fault
| Symptom | Likely cause | What to check or change |
|---|---|---|
| No connection | Wrong mode, address or port; unreachable listener; blocked UDP traffic. | Confirm caller targets the listener’s correct reachable address and port. Check the listener’s local bind, host firewall and router/NAT UDP forwarding. Verify both endpoints use compatible connection roles. |
| Connection fails when encryption is enabled | Incompatible passphrase or key settings; enforced encryption rejects the peer. | Compare the encryption configuration at both endpoints and correct the mismatch without exposing the passphrase. |
| Connection establishes but the receiver cannot use the feed | The endpoints do not agree on a compatible media format, or the source/encoder is not producing the expected stream. | Test the source and encoder separately, then confirm the receiving application supports the supplied media format. |
| Playback breaks up or packets are not recovered in time | Latency buffer may be too short for RTT variation, or the path may have loss or insufficient bandwidth. | Watch RTT and connection statistics. Test a longer receiver latency, then reassess delay and recovery. Check available bandwidth and packet loss before adding FEC or bonding. |
| Performance varies over time | RTT or link quality is changing, particularly on wireless or mobile paths. | Monitor the live connection rather than relying on a single RTT reading. Retune latency against observed variation and confirm the link can sustain the stream. |
When connection setup fails, investigate reachability and matching modes before changing codecs or increasing bitrate; picture-quality changes will not repair a blocked UDP path.
Choose software or an SRT hardware encoder
A software workflow may already provide capture and encoding. A hardware workflow might use a frame grabber or capture device feeding a live encoder, or a network encoder that accepts the source format and supports SRT. A generic capture card is not automatically an SRT encoder.
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Before buying hardware, confirm in the manufacturer’s documentation that the specific model supports SRT and your intended receiver. Check supported codecs and resolutions, network interfaces, caller/listener/rendezvous options and encryption controls. No specific model, current listing, price or availability is established here.
Or let it run in the cloud
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Quick Recap
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