In the INET framework, you can indeed simulate multiple hosts (nodes) streaming UDP packets to a central Ground Control Station (GCS). However, you need to set up your simulation configuration correctly to achieve this.

Here's how you can set it up:

Here's a general idea of how you can calculate throughput in your application code:

• Keep track of the number of bytes sent by the UDPVideoStreamSvr application.
• Keep track of the time elapsed.
• Calculate throughput as bytes sent divided by time elapsed.

Remember to adapt your code according to your specific simulation requirements.

Lastly, ensure that your OMNeT++ environment is set up correctly, and your simulation network topology and configurations are accurate. Debugging and testing different scenarios are essential to get the desired results in network simulations.

To establish UDP video streaming between multiple nodes and a Ground Control Station (GCS), you can follow these steps:

Here is an example of how to establish UDP video streaming between multiple nodes and a GCS using the RTP protocol:

# On each node: import socket import rtp # Create a UDP socket socket = socket.socket(socket.AF_INET, socket.SOCK_DGRAM) # Bind the socket to a port socket.bind(('192.168.1.100', 5004)) # Create an RTP session rtpSession = rtp.RtpSession() # Start streaming video while True: # Get the next video frame videoFrame = getVideoFrame() # Encode the video frame rtpPacket = rtpSession.encode(videoFrame) # Send the RTP packet to the GCS socket.sendto(rtpPacket, ('192.168.1.200', 5005)) # On the GCS: import socket import rtp # Create a UDP socket socket = socket.socket(socket.AF_INET, socket.SOCK_DGRAM) # Bind the socket to a port socket.bind(('192.168.1.200', 5005)) # Create an RTP session rtpSession = rtp.RtpSession() # Start receiving video while True: # Receive an RTP packet from a node rtpPacket, senderAddress = socket.recvfrom(1024) # Decode the RTP packet videoFrame = rtpSession.decode(rtpPacket) # Display the video frame to the user displayVideoFrame(videoFrame)

This is just a basic example, and you may need to modify it to meet your specific needs. For example, you may need to add support for multiple video streams, or you may need to use a different video streaming protocol.

Considerations for UDP video streaming:

• UDP is a connectionless protocol, which means that there is no guarantee that packets will be delivered in order or that all packets will be delivered. This can make UDP video streaming less reliable than TCP video streaming.
• UDP is a lightweight protocol, which means that it has less overhead than TCP. This makes UDP video streaming more efficient than TCP video streaming, especially for high-bandwidth video streams.
• UDP video streaming is more susceptible to packet loss and jitter than TCP video streaming. This can cause the video to be choppy or to have dropped frames.

Overall, UDP video streaming is a good choice for applications where latency is more important than reliability, such as real-time video surveillance. However, it is important to be aware of the limitations of UDP before using it for video streaming.

To find the elapsed time in your case, you can keep track of the time at which the first packet was sent and the time at which the last packet was received. You can then calculate the difference between these two times to get the elapsed time.

Here’s an example of how you can calculate the elapsed time in C++ using the INET framework:

#include

using namespace omnetpp;

class MyModule : public cSimpleModule

{

private:

simtime_t startTime;

simtime_t endTime;

protected:

virtual void initialize() override;

virtual void handleMessage(cMessage *msg) override;

virtual void finish() override;

};

Define_Module(MyModule);

void MyModule::initialize()

{

startTime = simTime();

}

void MyModule::handleMessage(cMessage *msg)

{

// Handle incoming packets here

if (/* Last packet has been received */)

{

endTime = simTime();

}

}

void MyModule::finish()

{

simtime_t elapsedTime = endTime - startTime;

EV_INFO