Stream Settings

Select the Audio Encoder to assign to the stream.

• None (no content source selected)
• Audio Encoder 0 .. 15

(Only available if KLV or CoT has been installed) To enable metadata, select one of the defined inputs.

• (None)
• Select from list of defined metadata sources, e.g., HD-SDI-BNC-1, HD-SDI-BNC-2, HD-SDI-BNC-3, or HD-SDI-BNC-4

(Optional - only available as a licensed feature) To enable ad insertion, select one of the following from the SCTE 35 drop down menu:

• (None)
• Select from list of defined ad insertion sources, e.g., HD-SDI-BNC-1, HD-SDI-BNC-2, HD-SDI-BNC-3, or HD-SDI-BNC-4
  

For more information, see Setting Up Streams and Configuring Ad Insertion Sources

Data Carriage

Select the Protocol Type for the encoded stream.

• TS over UDP: MPEG2 transport stream over UDP (no RTP header)
• TS over RTP: MPEG2 transport stream over RTP
• TS over SRT: Secure Reliable Transport. See Configuring Secure Reliable Transport (SRT).

(Optional) Program Identifier used in the Program Map Table (PMT) of the TS stream. 0..65535

Enter the destination IP address in dotted-decimal format.

Note

The Multicast address range is from 224.0.0.0 to 239.255.255.255. Multicast addresses from 224.0.0.0 to 224.0.0.255 are reserved for multicast maintenance protocols and should not be used by streaming sessions. We recommend that you use a multicast address from the Organization-Local scope (239.192.0.0/14).

Enter the destination UDP port(s). Enter a number in the range 1025..65,535. Note that RTP streams use even numbers only within this range.

Controls the timing characteristics of packets transmitted on the network. Select either:

• VBR (Variable Bitrate): The stream is not controlled and packets are transmitted as they become available.
• CVBR (Capped Variable Bitrate): The maximum stream bitrate transmitted is limited by the Bandwidth Overhead parameter.
• CBR (Constant Bitrate): The stream bitrate, packet timing, and audio/video timing comply with MPEG-2 TS CBR definition. The maximum stream bitrate transmitted is limited by the Bandwidth Overhead parameter. If no data is available, idle cells may be inserted.

Note

Timing & Shaping settings combine and replace the Traffic Shaping, Idle Cells and Delayed Audio parameters from the Makito X:

• VBR: Shaping=Off, Idle Cells=Off, Delayed Audio=Off
• CVBR: Shaping=On, Idle Cells=Off, Delayed Audio=Off
• CBR: Shaping=On, Idle Cells=On, Delayed Audio=On

(CBR/CVBR streams) Specifies the maximum stream bandwidth overhead that can be used for lost packets recovery.
Range = 5–50%

TTL

(Time-to Live for stream packets) Specifies the number of router hops that IP packets from this stream are allowed to traverse before being discarded.
Range = 1..255

ToS

(Type of Service) Specifies the desired quality of service (QoS). This value will be assigned to the Type of Service field of the IP Header for the outgoing streams.
Range = 0..255 (decimal) or 0x00..0xFF (hex)

Important

A DiffServ or DSCP (Differentiated Services Code Point) value must be converted to a ToS precedence value. For example, AF41 or DSCP 34 becomes ToS 136. For more information, see RFC2474.

Note

The ToS setting must be chosen so as to not interfere with Voice over IP systems and other equipment that may reside on your network. For example, when the ToS value for a stream is set to 0xB8, it can interfere with some third party Voice / IP Telephony systems.

Specifies the type of FEC matrix scheme.

• Check this checkbox to align the FEC blocks in the matrix structure (i.e., sequential columns within a group start on the same row).
• If left unchecked, the blocks are a staggered series of FEC packets (i.e., each column starts on the row below the row on which the previous column started).

Note

The enabled Block Aligned setting corresponds to the aligned FEC discussed in Annex C of SMPTE 2022-1.

The disabled Block Aligned setting corresponds to the non-aligned FEC discussed in Annex B of SMPTE 2022-1.

Specifies the SRT Connection Mode:

• Caller: The SRT stream acts like a client and connects to a server listening and waiting for an incoming call.

• Listener: The SRT stream acts like a server, and listens and waits for clients to connect to it.

  

  Note

  The firewall must be configured to allow incoming Caller connections to reach the Listening device.

• Rendezvous: Allows calling and listening at the same time.

  

  Note

  To simplify firewall traversal, Rendezvous Mode allows the encoder and decoder to traverse a firewall without the need for IT to open a port, but requires that the firewall not remap the UDP port for the stream.

(Caller and Rendezvous modes) Specifies the destination IP address for the SRT stream, or for Path 1 and Path 2 if Path Redundancy is configured.

Tip

You can also enter a Fully Qualified Domain Name (FQDN).

(Caller and Rendezvous modes) Specifies the UDP source port for the SRT stream, or for Path 1 and Path 2 if Path Redundancy is configured. If not filled in, a (default) source port will be assigned.

Note

This simplifies firewall configuration as the firewall/NAT rules can be precisely tailored to the SRT stream.

Specifies the SRT receiver buffer that permits lost packet recovery. The size of this buffer adds up to the total latency. A minimum value must be 3 times the round-trip-time (RTT). Range = 20–8000 ms

Note

Latency is for the SRT protocol only and does not include the capture, encoding, decoding and display processes of the end-point devices.

The SRT buffer, configured as "Latency'", is the time reserved in the decoder to recover missing packets.

(Only available if Encryption is enabled) Configures Authenticated-Encryption with Associated-Data (AEAD). AEAD not only requires a key (passphrase) to decipher the encrypted stream, but also binds it to certain associated data. This associated data (composed of some non-confidential information about the stream) is integrity-protected, which allows users to determine if their streams have been tampered with or not. 

None - Default for Caller and Rendez-vous modes. Not available for Listener mode.

Auto - Default for Listener mode. Not available for Caller or Rendez-vous modes.

AES-GCM - Available for Caller, Listener, and Rendez-vous modes.

Note

Setting the authentication to Auto allows the Listener to accept Callers with authentication set to either None or AES-GCM. Listeners with the authentication set to AES-GCM will only accept Callers which are also set to AES-GCM.  

(Only required and accepted if Encryption is enabled) Specifies a string used to generate the encryption keys to protect the stream.
Range = 10–79 UTF8 characters 

(SRT streams only) Specifies the maximum stream bandwidth overhead that can be used for lost packets recovery.
Range = 5–50%

Note

SRT streams may temporarily overshoot the defined bandwidth overhead limit.

(SRT Caller only) Select the format to configure the Stream Publishing ID:  

• Standard Keys: Simplifies defining the Stream Publishing ID. The Stream Publishing ID  field is read-only and auto-fills when you fill in the  Resource Name  and  User Name  fields.
  Example Standard Keys format Stream ID: #!::u=admin,r=haivision1 ,m=publish

• Custom: Select Custom to enter the Stream Publishing ID using your own format. 

  

  Tip

  If you first select Standard Keys format and fill in the  Resource Name  and  User Name fields, you can then modify or complete the resulting  Stream Publishing ID by switching to Custom format.

  See  Configuring SRT Access Control.

Enter the SAP advertising UDP port. Default=9875.

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