Connect an SRT stream as an input

Secure Reliable Transport (SRT) is a low-latency protocol used to stream over the Internet to SRT destinations (decoders), such as CDNs and other SRT decoders.

You can configure Unify as an SRT destination (decoder) and ingest SRT streams as video inputs that you can add to custom layouts for switching, recording, and streaming.

When adding an SRT input, be sure to select the correct SRT mode. For example, if the SRT source stream is using rendezvous mode, ensure you set the SRT input in the Admin panel to rendezvous mode. If the SRT source stream is using caller mode, then set the SRT input in the Admin panel to listener mode. For more information about SRT modes, see About streaming using SRT.

What's needed for this setup

• The IP address and SRT port of the SRT source (encoder) is needed for caller mode and rendezvous mode. For Unify, this must be the Public IP v4 address as IP V6 is not supported.

• For rendezvous mode, you must configure the SRT input to use the same port value as the SRT source. For an example SRT stream using rendezvous mode, see Set up an SRT stream using rendezvous mode.

• For caller/listener modes, you must know which mode the SRT source uses and configure the SRT input on Unify to use the corresponding mode.

• Caller mode can be used only for connection with the publicly accessed destination/source.

• Listener mode can be used for connection with any source/destination (including within a local network behind the firewall).

• For example, if the SRT source is in caller mode, then the input must be in listener mode. For an example SRT stream using caller/listener modes, see Set up an SRT stream using caller and listener modes.

• Unify supports 128 bits, 192 bits, and 256 bits AES encryption. Special characters are not supported in the passphrase.

If AES encryption and a passphrase is configured, ensure this information is provided to the SRT source. The SRT stream from the source must use the same AES and passphrase values, see Set AES encryption and a passphrase for SRT.

Add an SRT stream as an input using the Admin panel

• Log in to the Admin panel as admin, see Connect to the Admin panel.

• From the Inputs menu, select Add SRT input. The add input page opens.

• In the Connection mode field, select the SRT mode:

• For Caller mode, enter the following:

• *URL *- the URL of the SRT source (encoder) using the format *srt://**<**encoder-ip-address**>*:<**port**>, where the <**encoder-ip-address**> and <**port**> could be a Pearl device sending an SRT stream. For Unify, this must be the Public IP v4 address or fully qualified domain name of the source, because IP V6 is not supported.

• Source port - enter the port number for Unify to use for SRT traffic.

• Stream ID - In the Stream ID field, enter a unique name for this stream.

• For Listener mode, optionally enter the following:

• Enter the Port number for this device to use for SRT traffic.

• For **Rendezvous **mode enter the following:

• *URL *of the SRT source (encoder) using the format *srt://**<**encoder-ip-address**>*:<**port**>, where the <**encoder-ip-address**> and <**port**> values are provided by the SRT encoder device.

• Enter the following optional settings:

• Encryption - select the check box and choose the AES Key length if the SRT destination requires AES encryption. These settings must match the encryption settings of the destination device.

• Passphrase - Enter a passphrase if one is required along with AES encryption.

• Latency - 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.If you require real-time responses from your audience, you should use as little latency as possible.

• Audio delay - The audio delay feature helps you adjust audio coming from external sources so that your video and audio sources are synchronized when captured, streamed, and recorded. You can apply from -300 ms to +300 ms of delay. By default, there's 0 ms of delay applied to audio sources.

• Disable color range conversion - Color range conversion ensures that the ingested SRT video signal is full-range RGB (0-255), even if the signal was sent with limited RGB color range. By default, color range conversion is enabled.

• No signal image - Select an image to be displayed if there is no signal in the SRT stream. from the dropdown list. The DEFAULT image displays the words NO SIGNAL. To add a new image, select Media in the Configuration menu and click Choose File.

• Timeout - Select an amount of time with no signal before the No signal image appears.

• Click Apply.

We recommend using the default latency value for an initial test stream. After you perform a test stream, you can determine the Quality of Service (QoS) and calculate the latency requirements. To set the SRT stream latency, see Adjust latency and view the SRT stream status.