An SRT server is the managed contribution endpoint. Configure latency and bandwidth for the real network, secure it, then use telemetry and controlled routes rather than guessing from player symptoms.
開始前の準備
A reachable UDP port and a firewall rule limited to the required sources when possible.
A real SRT encoder or test source and a receiver such as VLC or a downstream Callaba module.
Measured network RTT and expected peak bitrate.
設定の説明
運用者に必要なコントロールだけを説明し、内部フィールド名や実装イベントは表示しません。
Listener
Identity and network capacity of the SRT endpoint.
- Name and active state
Operator label and lifecycle state of this SRT server.
- Listener port
UDP port used by incoming SRT connections.
- Receiver port
Optional distinct port used by the receiver side of the configured workflow.
- Latency
SRT recovery buffer sized for the measured network path.
Start from measured RTT and test under representative packet loss.- Maximum bandwidth
Headroom SRT may use for retransmission relative to the media bitrate.
- Receive buffer
Host buffer available to the SRT receiver.
- Connection timeout
How long an unavailable peer may remain unresolved before the session is treated as failed.
Security and access
Protect media and restrict who may publish or receive.
- Passphrase
Optional SRT encryption secret shared by authorized peers.
Use a strong secret and distribute it separately from the URL.- Access mode
Require explicit Stream access objects, restrict by host, or deliberately allow broader access.
- Publisher and receiver roles
Separate clients that send media from clients that consume it.
Routing and recovery
Connect this endpoint to one or more SRT peers.
- Routing mode
Disable routing or use Callaba/public-address peer definitions.
- PUSH or PULL
PUSH sends toward a listener; PULL connects to an available source.
- Routing hosts
Ordered peer addresses used by the configured route workflow.
Observability
Use live evidence before changing transport parameters.
- Statistics interval
Frequency used for supported statistics collection or callbacks.
- Live bitrate
Shows whether media is currently arriving.
- Network RTT
Round-trip time reported by the SRT session.
- Connection and event history
Session and transport events used to diagnose disconnects.
安全な初回ワークフロー
- 01
Create the server with an unused UDP port and a clear operator name.
- 02
Measure the real path, then set latency, maximum bandwidth and timeout for that network.
- 03
Add a passphrase and explicit publisher/receiver Stream access.
- 04
Start the server and connect one real publisher.
- 05
Connect one receiver or downstream module and confirm live bitrate plus Network RTT.
- 06
Only after the single path is stable, add routing or a backup PULL source.
ワークフロー例
Secure remote contribution into production
A field encoder sends a feed over the public internet.
構築手順
- 1
Create the SRT listener and size latency from the measured path.
- 2
Set a passphrase and create a publisher Stream for the encoder.
- 3
Connect the encoder and confirm bitrate, RTT and stable session state.
- 4
Select the verified SRT source in Multiview, Recording or Restreaming.
Primary and backup SRT contribution
Production has two independently reachable feeds and needs a controlled recovery path.
構築手順
- 1
Add the independently tested primary and backup hosts to the PULL routing plan.
- 2
Verify each source alone before enabling the ordered route workflow.
- 3
Watch both source health states in Multiview or the operator interface.
- 4
For planned switching, use the preferred-route control; for reconnect recovery, observe the active session and output.
- 5
Simulate a source loss during a maintenance window and record recovery behavior.
Multilingual browser event from SRT feeds
Production already provides one SRT feed per language.
構築手順
- 1
Receive and validate one independent SRT source for every produced language.
- 2
Create one Web player per verified language feed and name it for the audience.
- 3
Add those players to a Language list group and choose the default language.
- 4
Test switching, access rules and playback in a clean browser session.
結果を確認
- A real publisher and receiver remain connected for the planned test period.
- Live bitrate confirms media arrival and Network RTT remains within the tested latency budget.
- Unauthorized stream identities or hosts are rejected by the selected access policy.
- A backup-source test produces the documented recovery behavior.
トラブルシューティング
Session connects but media breaks up
確認項目- Compare bitrate to available upload.
- Check RTT and packet-loss behavior.
- Review latency and maximum bandwidth.
Increase recovery headroom only from measured evidence; also correct the source bitrate or network bottleneck.
Publisher cannot connect
確認項目- Check UDP firewall and public address.
- Match port, mode, passphrase and Stream identity.
Test one publisher without downstream modules, then restore the verified receiver path.
The backup does not take over as expected
確認項目- Verify both sources independently.
- Review route order, mode and timeout.
- Observe the active session during failure.
Correct the failed source or routing definition and repeat a controlled failure test.