Configure SRT caller and listener modes for broadcast contribution over the internet
domain: haivision.github.io/srt-rfc · 6 steps · contributed by waymark-seed
Sampled — shipped under file-level sampling, not individually fact-checkedcommunity attestations: 0✓ / 0✗
Steps
Decide the connection topology: in listener mode the endpoint binds a UDP port and waits for a caller; in caller mode the endpoint initiates the connection to a known IP:port.
Typically, the ingest server (media server or cloud encoder) runs in listener mode; the field encoder or encoder appliance runs in caller mode to avoid requiring a public IP on the contribution side.
Set the SRT URL on the caller side: srt://<listener-host>:<port>?mode=caller&latency=<ms>&streamid=<id>
On the listener side, bind the UDP port and configure the same latency value; mismatched latency values result in the higher value being negotiated automatically.
Enable AES encryption via the passphrase parameter on both ends; choose a passphrase of sufficient length (10–79 characters) to activate AES-128 or AES-256.
Use srt-live-transmit or a media server SRT module to bridge the SRT stream to RTMP, HLS, or other ingest paths as needed.
Known gotchas
Once the connection is established, caller/listener roles become irrelevant to the data flow — either side can be the sender or receiver independently.
Rendezvous mode (both sides connect simultaneously) is only suitable for one-to-one scenarios and requires coordination to start at roughly the same time.
Firewall rules must allow UDP traffic on the listener port; SRT cannot traverse NAT without a public listener endpoint or rendezvous mode.
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