Radiant Communications VL4500SRT Gateway User manual

VL4500SRT GATEWAY SERIES

Installation and Operations Manual

User Guide

Radiant

Communications

Corporation

V L 4 5 0 0 S R T G A T E W A Y S E R I E S

Installation and Operation Manual

 Radiant Communications Corp. 2017

5001 Hadley Road

South Plainfield, NJ 07080

Phone 908.757.7444 • Fax 908.757.8666

www.rccfiber.com

 
 
Table of Contents 
 
Product Summary .................................................................... 1 
Configuring an SRT Stream ............................................................ 2 
SRT Parameters ............................................................................. 6 
Installation ................................................................................ 9 
Communication ...................................................................... 10 
Network Management ................................................................... 10 
Configuration .......................................................................... 11 
SRT Settings ................................................................................. 13 
SRT Statistics ............................................................................... 15 
Network Settings ........................................................................... 18 
System Setup ................................................................................ 20 
Firmware Upgrade: ................................................................................. 21 
User Presets ................................................................................. 22 
Revisions ............................................................................... 23 

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Product Summary

SRT protocol is a transport technology that optimizes streaming performance across

unpredictable networks like the Internet with secure streams and easy firewall traversal,

bringing the best quality live video over the worst networks. It accounts for packet loss, jitter,

and fluctuating bandwidth, maintaining the integrity and quality of your video.

Internet streaming obstacles are:

1. Packet Loss - Packets being discarded by routers

2. Jitter - Packets arriving at different times than expected and sometimes out of order

3. Latency - Time from sender to receiver

4. Bandwidth - The fluctuating capacity between points

The SRT protocol relies on bi-directional UDP traffic to optimize video streaming over public

networks. In addition to the video data that is sent in one direction - from a content source

device (such as a VL4500 Encoder) to a Destination (such as a VL4500 SRT Gateway) - there is a

constant exchange of control information between the two endpoints, including “keep alive”

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packets approximately every 10 ms, which allow SRT streams to be automatically restored after

a connection loss.

VL4500

SRT

INTERNET

VL4500SRT

SRT Gateway

MPEG TS

MULTICAST

FIREWALL FIREWALL

SDI

CVBS

SOURCE DESTINATION

Configuring an SRT Stream

Along with the standard parameters associated with any streaming output, there are other

important values you must specify for an SRT stream. To introduce you to these parameters,

let’s take a look at a hypothetical example of an SRT stream being sent to a destination device,

and see what happens over time.

The diagram below depicts a stream being sent over a channel of some kind, such as a LAN or

Internet connection, with a certain capacity. Packets being sent from a source are stored at the

destination in a buffer. Let’s say at some point there is a total link failure, and then, shortly after,

the connection is re-established. So, for a short period of time, the destination is receiving no

data.

SRT deals with such situations in two ways. The first is that the destination relies on its buffer to

maintain the stream output at its end. The size of this buffer is determined by the SRT Latency

setting. Once the link is re-established, the source is able to resume sending packets, including

re-sending the packets lost during the link failure. To handle this extra “burst” of packets, an SRT

stream allows for a certain amount of overhead. This Bandwidth Overhead is calculated such

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that, in a worst-case scenario, the source can deliver the number of packets “lost” during the

link failure (area A) over a “burst time” (area B), where area B must be equal to area A. The

maximum time period for which a burst of lost packets can be sustained without causing an

artifact is:

SRT Latency (ms) * Bandwidth Overhead (%) ÷ 100

With your source and destination devices set up (including having established call modes and

any firewall settings), follow these steps to configure an SRT stream:

1. Measure the Round Trip Time (RTT) using the ping command.

root@VL4500:~# ping 192.168.16.245

PING 192.168.16.245 (192.168.16.245): 56 data bytes

64 bytes from 192.168.16.245: seq=1 ttl=64 time=0.162 ms

In the above example tested response (RTT = 0.162 ms)

• If ping does not work or is not available, set up a test SRT stream and use the RTT

value from the VL4500 SRT Gateway Statistics page.

• If the RTT is <= 20 ms, then use 20 ms for the RTT value. This is because SRT

does not respond to events on time scales shorter than 20 ms.

2. Measure the Packet Loss Rate.

A channel’s Packet Loss Rate drives the SRT Latency and Bandwidth Overhead calculations.

This loss rate can be extracted from iperf stats.

Connect to the VL4500SRT gateway via telnet or serial port and start iperf server typing the

command iperf -su -I 1

Connect to the VL4500 encoder via telnet or serial port and start iperf client typing the

command iperf -c <VL4500SRT IP ADDRESS> -b <DATARATE> -i 1 -t 20

-b Setting UDP bandwidth (-b)

-i Reporting interval in seconds

-t Length of test in seconds

Below is a test performed by VL4510 Encoder with IP 192.168.16.249 and VL4500SRT

Gateway with IP 192.168.16.245. Test is for UPD only transfer at 38M data rate at 1sec

interval for 20 seconds.

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root@VL4500:~# iperf -su -i 1

------------------------------------------------------------

Server listening on UDP port 5001

Receiving 1470 byte datagrams

UDP buffer size: 106 KByte (default)

------------------------------------------------------------

[ 3] local 192.168.16.245 port 5001 connected with 192.168.16.249 port 60253

[ ID] Interval Transfer Bandwidth Jitter Lost/Total Datagrams

[ 3] 0.0- 1.0 sec 4.54 MBytes 38.1 Mbits/sec 0.017 ms 8/ 3248 (0.25%)

[ 3] 1.0- 2.0 sec 4.54 MBytes 38.1 Mbits/sec 0.012 ms 0/ 3236 (0%)

[ 3] 2.0- 3.0 sec 4.54 MBytes 38.1 Mbits/sec 0.001 ms 0/ 3237 (0%)

[ 3] 3.0- 4.0 sec 4.54 MBytes 38.1 Mbits/sec 0.004 ms 0/ 3236 (0%)

[ 3] 4.0- 5.0 sec 4.54 MBytes 38.1 Mbits/sec 0.001 ms 0/ 3236 (0%)

[ 3] 5.0- 6.0 sec 4.54 MBytes 38.1 Mbits/sec 0.002 ms 0/ 3237 (0%)

[ 3] 6.0- 7.0 sec 4.54 MBytes 38.1 Mbits/sec 0.001 ms 0/ 3236 (0%)

[ 3] 7.0- 8.0 sec 4.54 MBytes 38.1 Mbits/sec 0.001 ms 0/ 3236 (0%)

root@VL4500:/opt/rcc# iperf -c 192.168.16.245 -b 38M -i 1 -t 20

WARNING: option -b implies udp testing

------------------------------------------------------------

Client connecting to 192.168.16.245, UDP port 5001

Sending 1470 byte datagrams

UDP buffer size: 106 KByte (default)

------------------------------------------------------------

[ 3] local 192.168.16.249 port 60253 connected with 192.168.16.245 port 5001

[ ID] Interval Transfer Bandwidth

[ 3] 0.0- 1.0 sec 4.54 MBytes 38.1 Mbits/sec

[ 3] 1.0- 2.0 sec 4.54 MBytes 38.1 Mbits/sec

[ 3] 2.0- 3.0 sec 4.54 MBytes 38.1 Mbits/sec

[ 3] 3.0- 4.0 sec 4.54 MBytes 38.1 Mbits/sec

[ 3] 4.0- 5.0 sec 4.54 MBytes 38.1 Mbits/sec

[ 3] 5.0- 6.0 sec 4.54 MBytes 38.1 Mbits/sec

[ 3] 6.0- 7.0 sec 4.54 MBytes 38.1 Mbits/sec

[ 3] 7.0- 8.0 sec 4.54 MBytes 38.1 Mbits/sec

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3. Using the following table*, find the RTT Multiplier and Bandwidth Overhead

values that correspond to your measured Packet Loss Rate:

Worst Case

Loss Rate (%)

RTT

Multiplier

Bandwidth

Overhead (%)

Minimum SRT Latency

(for RTT <= 20 ms)

≤1

3

33

60

≤3

4

25

80

≤7

5

20

100

≤10

6

17

120

4. Determine your SRT Latency value using the following formula:

SRT Latency = RTT Multiplier * RTT

• If RTT < 20, use the Minimum SRT Latency value in the table above.

6. Determine the stream bitrate.

• The steam bitrate is the sum of the video, audio and metadata essence bit rates, plus

an SRT protocol overhead. It has to respect the following constraint:

Tested Bandwidth > Stream Bitrate * (100 + Bandwidth Overhead) ÷ 100

• If this is not respected, then the video/audio/metadata bitrate must be reduced until

it is respected.

• It is recommended that a significant amount of headroom be added to cushion

against varying channel capacity, so a more conservative constraint would be:

0.75 * Tested Bandwidth > Stream Bitrate * (100 + Bandwidth Overhead) ÷ 100

7. Configure Firewalls.

Most common application scenario is streaming using Caller and Listener mods.

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192.168.1.245

SRT/UDP port 51012

192.168.16.1

NAT

Packet Filtering

174.10.10.1

FIREWALL B

VL4500 Encoder – SRT Source

MEDIA

CONTROL

173.10.10.1

Packet Filtering

NAT

192.168.1.1

FIREWALL A

RESPOSE

INTERNET

MEDIA

CONTROL

RESPOSE

192.168.16.245

SRT/UDP port 51011

VL4500SRT Gateway

SRT Destination

Firewall Setting

Value

Description

Protocol

UDP

SRT is based on UDP protocol

Source IP

192.168.1.245

The VL4500 Encoder IP address

WAN IP

173.10.10.1

Public IP address of the router

Port Forward 1

51012 → 192.168.1.245

Incoming response messages port forward

Outbound NAT

Port 51011

Set Outbound NAT Port

Firewall Setting

Value

Description

Protocol

UDP

SRT is based on UDP protocol

Source IP

192.168.16.245

The VL4500SRT Gateway IP address

WAN IP

174.10.10.1

Public IP address of the router

Port Forward 1

51011 → 192.168.16.245

Incoming media/control messages port forward

Outbound NAT

Port 51012

Set Outbound NAT Port

SRT Parameters

Round Trip Time (RTT) is the time it takes for a packet to travel from a source to a destination

and back again. It provides an indication of the distance (indirectly, the number of hops)

between endpoints on a network. Between two SRT devices on the same fast switch on a LAN,

the RTT should be almost 0. Within the Continental US, RTT over the Internet can vary

depending on the link and distance, but can be in to 60 to 100 ms range. Transoceanic RTT can

be 60-200 ms depending on the route. RTT is used as a guide when configuring Bandwidth

Overhead and Latency. To find the RTT between two devices, you can use the ping command.

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RTT Multiplier is a value used in the calculation of SRT Latency. It reflects the relationship

between the degree of congestion on a network and the Round Trip Time. As network

congestion increases, the rate of exchange of SRT control packets (as well as retransmission of

lost packets) also increases. Each of these exchanges is limited by the RTT for that channel, and

so to compensate, SRT Latency must be increased. The factor that determines this increase is

the RTT Multiplier, such that:

SRT Latency = RTT Multiplier * RTT

The RTT Multiplier, then, is an indication of the maximum number of times SRT will try to resend

a packet before giving up.

Packet Loss Rate is a measure of network congestion, expressed as a percentage of packets lost with

respect to packets sent.

Constant loss refers to the condition where a channel is losing packets at a constant rate. In such cases,

the SRT overhead is lower bound limited, such that:

Minimum Bandwidth Overhead = 1.65 * Packet Loss Rate

Burst loss refers to the condition where a channel is losing multiple consecutive packets, up to the

equivalent of the contents of the SRT latency buffer. In such cases, the SRT overhead is lower bound

limited, such that:

Minimum Bandwidth Overhead = 100 ÷ RTT Multiplier

Burst losses that last longer than the SRT Latency will result in stream artifacts. SRT Latency should

always be set to a value above your worst case burst loss period.

Bandwidth Overhead

The control packets associated with an SRT stream do, of course, take up some of the available

bandwidth, as do any media packet retransmissions. When configuring an SRT stream, you will need

to specify a Bandwidth Overhead value to allow for this important factor.

The portion of audio and video content in the stream is determined by their respective bit rate settings,

which are configured on the audio and video encoders themselves. SRT Bandwidth Overhead is

calculated as a percentage of the A/V bit rate, such that the sum of the two represents a threshold bit

rate, which is the maximum bandwidth the SRT stream is expected to use.

The SRT Bandwidth Overhead is a percentage you assign, based in part on the quality of the network

over which you will be streaming. Noisier networks will require exchanging more control packets, as

well as resending media packets, and therefore a higher percentage value.

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SRT Bandwidth Overhead should not exceed 50%. The default value is 25%.

Let’s say you are streaming video at a bit rate of 1000 kbps, and audio at 128 kbps. This gives a total of

1128 kbps, which we will round up to 1200 kbps to account for any metadata and other ancillary data.

This is the Average Bandwidth, which is calculated automatically based on your actual output settings.

If you accept the default Bandwidth Overhead setting of 25%, then the total bandwidth reserved for

the SRT stream will be:

1200 + (25% * 1200) = 1500 kbps (1.5 Mbps)

This is the maximum bandwidth SRT will use. If there is no loss, only a slight overhead for control is

used. As long as this total SRT bandwidth is less than or equal to the bandwidth available between the

SRT source and destination devices, the stream should flow from one to the other without incident.

Latency is a time delay associated with sending packets over a (usually unpredictable) network.

Because of this delay, an SRT source device has to queue up the packets it sends in a buffer to

make sure they are available for transmission and re-transmission. At the other end, an SRT

destination device has to maintain its own buffer to store the incoming packets (which may

come in any order) to make sure it has the right packets in the right sequence for decoding and

playback. SRT Latency is a fixed value (from 20 to 8000 ms) representing the maximum buffer

size available for managing SRT packets.

An SRT source device’s buffers contain unacknowledged stream packets (those whose reception

has not been confirmed by the destination device).

An SRT destination device’s buffers contain stream packets that have been received and are

waiting to be decoded.

The SRT Latency should be set so that the contents of the source device buffer (measured

in msecs) remain, on average, below that value, while ensuring that the destination device

buffer never gets close to zero.

The value used for SRT Latency is based on the characteristics of the current link. On a fairly

good network (0.1-0.2% loss), a “rule of thumb” for this value would be four times the RTT. In

general, the formula for calculating Latency is:

SRT Latency = RTT Multiplier * RTT

SRT Latency can be set on both the SRT source and destination devices. The higher of the

two values is used for the SRT stream.

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Installation

The VL4500SRT chassis can be mounted in a standard EIA, 19” rack.

Secure the Encoder’s front panel to the rack by inserting four machine screws, with cup

washers, through the four mounting holes in the front panel.

When installing one or more encoders in a headend rack, it is recommended to leave a 1 rack

unit space (1.75” high) between units to maximize air flow.

To power up the unit, connect the 12VDC power supply to the unit.

The POWER LED on the front-panel will light.

Front Panel

SERIAL PORTPOWER

Radiant

Communications

Corporation

Series VL4500SRT

Rear Panel

POWER

+ V

TS OUT

EARTH

TS

TS OUT

MGMT

10/100/1000

VL4500SRT

10/100/1000

POWER

1 2

GND

+12

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Communication

The Encoder can be controlled and managed with either the out-of-band network port located on

the front panel or via the RS232 serial port. In-band management through the TS port is possible as

well.

Network Management

Initially, the simplest way to setup the networked PC to communicate with the VL4500SRT is to put the

PC on the same subnet as the VL4500. The default IP address of the VL4500SRT management port is:

192.168.1.253

NOTE: The static IP of the computer cannot be the same as the static IP of the VL4500.

The following steps explain how to change the IP address of a computer running windows 7:

1. On your computer, open the "Control Panel"

2. Click on “Network and Internet”

3. Click on the "View network status and tasks"

4. Click on “Change Adapter Settings” on left hand side of the window

5. Right-click on the "Local Area Connection", and then click on the "properties".

6. A dialog box entitled "Local Area Connection Properties" will appear. In this box, double-

click on the "Internet Protocol Version 4 (TCP/IPv4)".

7. A dialog box entitled "Internet Protocol Version 4 (TCP/IPv4) Properties" will appear.

8. Select the "Use the following IP address" option and enter the following addresses:

IP address: 192.168.1.2

Subnet mask: 255.255.255.0

9. Click OK to close the dialog box. Now your computer is ready to communicate with the

unit.

10. Open an Internet Explorer web browser on your computer and enter the following URL

http://192.168.1.253

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Configuration

The VL4500SRT Web management system is accessible through MGMT Ethernet interface.

Settings are grouped by tabs for:

• Video Settings

• Audio Settings

• TS Settings

• Network Settings

• System Settings

• User Presets

Note:

Changed settings will take effect only after the Encoder is started or restarted via the buttons on

the bottom of the screen.

Settings can be saved to a desired preset; preset includes also the current state of the Encoder

The encoder is being shipped with encoder factory preset. Saving on that preset is not possible. After

the unit is configured, a new preset should be created and saved.

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The top portion of the web based management screen provides information about part number,

firmware version, uptime and unit name.

Bottom portion provides control to Stop, Start, Restart (soft reboot) the encoder, as well as creating

new presets, changing presets and saving them.

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SRT Settings

SRT Mode:

1. Listener Mode Sets a device to wait for a request to start an SRT streaming session. The Listener

device only needs to know that it should listen for an SRT stream on a certain port.

2. Rendezvous mode allows two devices behind firewalls to negotiate an SRT session over a

mutually agreed upon port. Both source and destination must be in Rendezvous mode.

Input and Output Interfaces can be set to either Management or TS interface. Normal practice is to set

the Management data and incoming SRT on same port, connected via firewall to public network, and

UDP output on TS port.

Source port sets the listening SRT port, the destination port on SRT encoder must be the same as source

port on SRT gateway.

The VL4500SRT gateway supports both Unicast and Multicast IP delivery with user adjustable TTL.

Destination IP address, port and TTL need to be set.

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A Unicast transmission/stream sends IP packets to a single recipient on a network. A Multicast

transmission sends IP packets to a group of hosts on a network. If the streaming video is to be

distributed to a single destination, then the destination IP address and port on the VL4500SRT must

equal to the receiving decoder IP address and port. If the application requires the decoding of the

stream at multiple concurrent locations, then the destination IP address and port must be set to a valid

Multicast IP address in the range of 224.0.0.0 to 239.255.255.255.

Multicasting is not supported on some legacy network devices. Before using the VL4500SRT gateway

in Multicast streaming mode, check the functional specifications of your network infrastructure to

ensure that the Multicast stream will not create major traffic on your network. Verify that your

backbone switch supports Internet Group Messaging Protocol (IGMP) snooping, which allows the core

of your network to ignore the traffic streams that Multicasting may generate.

Latency, overhead and encryption need to be set with same values as the VL4500 encoder, please refer

to VL4500 Encoder Series manual for more details.

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SRT Statistics

The Statistics section provides an overview of an SRT stream’s condition.

Common Statistics:

Parameter

Description

Up Time

The length of time the stream is actively streaming

RTT

Round Trip Time (RTT) is the time required for a packet to travel from a VL4500 Encoder to a

specific VL4500SRT and back again.

Bandwidth

Estimated maximum bandwidth available as viewed from the VL4500SRT device.

Receiver Global Measurements

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Parameter

Description

Bitrate

The length of time the stream is actively streaming

Total Received

Packets

Round Trip Time (RTT) is the time required for a packet to travel from a VL4500 Encoder to a

specific VL4500SRT and back again.

Total Received

Bytes

Estimated maximum bandwidth available as viewed from the VL4500SRT device.

Total Lost Packets

Number of packets reported missing by the VL4500SRT.

Total Lost Bytes

Number of bytes reported missing by the VL4500SRT.

Total Skipped

Packets

Packets that have arrived at the VL4500SRT too late, or that never arrive at all. If the “time to play”

for a packet has passed, and it is either not at the VL4500SRT or arrives after the content it is

associated with has already played, that packet is reported as “skipped” on the VL4500SRT.

Usually this results in some type of video artifact (a replayed frame or video blocking).

Total Lost Bytes

Skipped Bytes, same as above.

Total ACK Sent

Total number of acknowledgment and feedback packets sent to the VL4500 encoder (this is a

measure of transmission progress).

Total NAK Sent

Total number of negative acknowledgment packets sent to the VL4500 encoder.

Total Undecrypted

Packets

Indicates whether or not the SRT stream is being successfully decrypted.

Total Undecrypted

Bytes

Indicates whether or not the SRT stream is being successfully decrypted.

Packets in Buffer

The amount stream packets received and waiting to processed.

Bytes in Buffer

Same as above.

Time in Buffer

Amount of time the transport packets stayed in the VL4500SRT buffer. This statistic shows the

portion of the VL4500SRT buffer up to the first missing packet (i.e. the time remaining to transmit

the missing packet before it’s too late). The value of the buffer in the absence of packet loss is just

below the latency value. In the presence of packet loss, it is between 0 and the latency value.

Buffer

Buffer size

Latency

A fixed value (from 20 to 8000 ms) representing the maximum buffer size available for managing

SRT packets.

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What to look for:

• A steadily increasing number of NAKs indicates a problem.

• It’s normal to see a few dropped packets, but there should not be too many. A low number of

dropped packets indicates you are using your bandwidth optimally.

• If the number of Skipped Packets increments slowly, the best thing to do is increase the SRT

Latency. If it increments in large jumps, the best thing to do is to lower the video bitrate, or to

increase the Bandwidth Overhead % if you have available capacity.

• Ideally, the time in buffer should be just below the latency value. If it goes to 0 often, then there

is most likely insufficient BW to support the desired bitrate (which should therefore be

lowered). If it goes occasionally to 0, then the SRT Latency should be increased.

• If the VL4500SRT shows too many skipped or dropped frames, increase the SRT Latency, lower

the video bit rate, and/or increase the Bandwidth Overhead %.

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Radiant Communications VL4500SRT Gateway User manual

Radiant Communications VL4500SRT Gateway User manual

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