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Adjust latency and view the SRT stream status

Secure Reliable Transport (SRT) achieves high-quality, low-latency streaming across unreliable Internet connections via UDP packets. If packets are lost in transit to the SRT destination, a request to retransmit the lost packets is sent back to Unify. Using the Admin panel, you can adjust the latency to improve the Quality of Service (QoS) of the stream and reduce the number of dropped packets.

During the SRT stream, you can view the stream statistics using the Admin panel and adjust the amount of latency based on the packet loss % and Round Trip Time (RTT). SRT stream statistics are provided on the streaming configuration page for a channel when Unify is configured as an SRT source. If Unify is configured as an SRT destination with an SRT input, then SRT statistics are available on the SRT input configuration page.

The following example shows SRT statistics for an SRT stream. The statistics section appears only while an SRT stream is active. It takes about 30 seconds for the statistics to appear after the SRT connection is established.

You can add from 80 ms to 8000 ms of latency to the SRT stream. Increasing latency gives more time to buffer packets and resend any that got lost in transit to the destination. If the latency value set for the stream is too low and there is packet loss over the network, retransmission of lost packets will not be possible and the stream quality will suffer.

Latency can be configured at the source and at the destination. SRT uses the highest of the two latency values.

The formula to calculate latency is:

SRT Latency = RTT Multiplier ´ RTT

where the recommended range of the RTT Multiplier is a value from 3 to 20.

The following table provides guidelines for what values to use when calculating latency. An RTT multiplier value less than 3 is too small for SRT to be effective and a value above 20 indicates a network with 100% packet loss. Ensure the measured buffer is less than or equal to the latency value you use.

Table 13 Suggested SRT latency values

Packet loss % RTT multiplier Recovery bandwidth overhead Minimum SRT latency (for RTT ≤ 20 ms)
% is ≤ 1 3 33 60
% is ≤ 3 4 25 80
% is ≤ 7 5 20 100
% is ≤ 10 6 17 120

These values are from the SRT Deployment Guide. For up-to-date calculations, visit www.srtalliance.org.

For example, if the % of packet loss is 0.53 and the measured RTT is 16.506 ms, the latency calculation is: 49.518 = 3 ´ 16.506 ms or 50 ms of latency (rounded up).

Adjust latency and recovery bandwidth overhead for an SRT stream using the Admin panel

The statistics section only appears while an SRT stream is active.

We recommend testing your settings. Start the SRT stream and use the stream statistics to evaluate the effect the latency and recovery bandwidth overhead values have on the packet loss % of the stream.

Set AES encryption and a passphrase for SRT

You can configure your Secure Reliable Transport (SRT) streams to use 128 bits, 192 bits, or 256 bits AES encryption using the Admin panel. You can also assign the SRT stream an alphanumeric passphrase if required. The encryption and passphrase settings on the SRT encoder sending the stream must match the settings on the SRT decoder receiving the stream.

The encryption key length is negotiated between the source and destination devices. The key length of the sender is used to determine the key length that is eventually used to secure the connection, and the receiver obtains the key length from the sender. As long as the passphrases match, the key length will be negotiated.

Set up AES encryption and a passphrase for an SRT stream using the Admin panel

Set up AES encryption and a passphrase for an SRT input using the Admin panel

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Source: unify-userguide.md · id unify/adjust-latency-and-view-the-srt-stream-status · Updated Oct 7, 2026 · view as markdown