Adding channels (connections)

Adding channels (connections)

Introduction

Quickstraem software can be used in many scenarios. Some examples are shown in use cases section. The easiest way to use Quickstream is to establish a point to point connection between two locations. In order to do it you need to have at least one Quickstream Node installed on the machine that has access to signal source. To manage a Quickstream Node, you have to configure Quickstream Client as described in the installation section.
Quickstream Node currently supports two connection methods 
  1. TS over UDP (UDP) 
  2. TS over SRT (SRT) 
Each configured channel may have a combination of UDP or SRT as inputs or outputs.

Transport stream over UDP protocol 

UDP as input 

  1. Type Unicast 
    1. Port - port number of the stream to receive (mandatory). An IP address is not required because, in Unicast type method, packets should use the IP address as the destination IP of the Network interface configured on the system 
    2. Network interface - if you select a network interface, the system will wait for UDP packets addressed to IP configured to that interface. If you choose the Automatic mode system routing table will be used to determine which interface should be selected  
  2. Type Multicast 
    1. IP Address - ipv4 address of the stream to receive
    2. Port - port number of the stream to receive (mandatory). An IP address is not required because, in Unicast type method, packets should use the IP address as the destination IP of the Network interface configured on the system
    3. Network interface - if you select the network interface, the system will send IGMP join request and wait for UDP packets on the selected interface. If you choose the Automatic mode system routing table will be used to determine which interface should be selected  

UDP as output (Unicast or Multicast) 

  1. IP Address - if you specify class D IP address from range 224.0.0.0/4, Quickstream Node will send packets as multicast packets to IP address provided as the destination. If you provide the IP address from A.B or C address space packets will be sent as unicasts. Provided IP Address will be used as the destination host IP address 
  2. Port - port number of the stream to be sent to
  3. Network interface - if you select a network interface, the system will use it to send packets. If you choose to stay with Automatic the system will automatically select an interface by reading the operating system routing table 

Transport stream over SRT protocol

Secure Reliable Transport (SRT) is an open-source protocol and technology stack designed for live video streaming over unmanaged IP networks such as the Internet. SRT is using UDP as an underlying transport layer and is using Automatic Repeat reQuest  (ARQ) techniques to fight with network errors whiie keeping latency low.
SRT works in three modes:
  1. Caller - in this mode, the system is actively initiating the connection to another system working in Listener mode. The caller must know the accessible/public IP address and port of the corresponding system 
  2. Listener - in this mode the system is waiting for connection on a specific port from another system working in Caller mode 
  3. Rendezvous -  in this mode, both systems should be in Rendezvous mode and each of them can initiate the connection to another one. 

Example configurations 



Point to point connection (Listener - Caller)

SRT as Input

  1. Caller Mode 
    1. IP Address - ipv4 address of Quickstream Node or any SRT enabled device 
    2. Port - destination IP that is open for connection. Can be the public IP address of SRT enabled device or redirected port on the firewall 
    3. Decrypt channel - if the received stream is encrypted the option should be enabled and decryption password should be entered 
    4. Receive latency - Specifies the SRT receiver buffer that permits lost packets to be recovered. Recommended latency is three times Round Trip Time (RTT). You can measure RTT by just sending ping between SRT sound and destination host. If the low-latency transmission is not required leave default latency of 4000 ms
  2. Listener Mode
    1. Port - Port to listen to incoming connections (from range 1024-65535)
    2. Network interface - interface and related IP address to bind. If Automatic is selected the system will wait for a connection on all interfaces 
    3. Decrypt channel - if the received stream is encrypted the option should be enabled and decryption password should be entered 
    4. Receive latency - Specifies the SRT receiver buffer that permits lost packets to be recovered. Recommended latency is three times Round Trip Time (RTT). You can measure RTT by just sending ping between SRT source and destination host. If the low-latency transmission is not required leave default latency of 4000 ms

SRT as output 

  1. Caller Mode 
    1. IP Address - ipv4 address of Quickstream Node or any SRT enabled device 
    2. Port - destination IP that is open for connection. Can be the public IP address of SRT enabled device or redirected port on the firewall (from range 1024-65535)
    3. Encrypt channel - if enabled, the channel will be encrypted with the provided password. The same password will need to be entered on the receiver side.  
    4. Receive latency - Specifies the SRT receiver buffer that permits lost packets to be recovered. Recommended latency is three times Round Trip Time (RTT). You can measure RTT by just sending ping between SRT sound and destination host. If the low-latency transmission is not required leave default latency of 4000 ms
  2. Listener Mode
    1. Port - Port to listen to incoming connections (from range 1024-65535)
    2. Network interface - interface and related IP address to bind. If Automatic is selected the system will wait for a connection on all interfaces 
    3. Encrypt channel -  if enabled, the channel will be encrypted with the provided password. The same password will need to be entered on the receiver side.  if the received stream is encrypted the option should be enabled and decryption password should be entered 
    4. Receive latency - Specifies the SRT receiver buffer that permits lost packets to be recovered. Recommended latency is three times Round Trip Time (RTT). You can measure RTT by just sending ping between SRT sound and destination host. If the low-latency transmission is not required leave default latency of 4000 ms

Establishing connections

If you want to establish connection between Quickstream Node A and Quickstream Node B perform the following steps:  
  1. Connect to Node A 
  2. Click the channels tab from the menu on the left 
  3. Click "Create new channel" 
  4. Enter name of the channel 
  5. Assign  previously added license to the channel
  6. Configure Input and Ouput 
  7. Click "Save channel" 
Repeat the same step on Node B. When finished click "Start" next to the channels you created to start.

Supported combinations of connected output and input



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