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License: MIT License
Multi user voice & messaging client and server. TCP session management. UDP communication.
License: MIT License
The client who's audio is transmitted shouldn't have their audio transmitted back to them.
Check if client connection ip == server client ip that packets get sent to.
Implement accounts, save login details for next time application is opened
Look into implementing VoIP/SIP.
Move logic for handling receive data into a worker thread. This free's up thread to receive new packets. Implement some sort of prepended identifier for handling different types of data.
Implement RTP
Add logs that write to disk for logging server output.
Show connection meter, displaying latency.
Maybe include timestamp in packet sent to measure connection
Refactor server to use thread for receiving multiple types of data.
Use TCP connection for changing username.
Broadcast changes to all clients
Make text chat work again.
Maybe look at fixing #9 before doing this so all data can go to same port.
When a client connects to server send all connected ServerClients to client. Then display there usernames in a list for user to see.
Needs a better UI design
Implement UDP video chat
Right now there is a datagram socket on port 3000 for the initial packet that tells the server a new client has connected, and another socket on port 3001 for when audio gets sent.
Merge them into one socket, include meta data with each packet, and use case statement to handle how each meta data type is handled.
Example:
Initial client connection packet should include /c/
Then on the server if packet data contains /c/. Do new connection workflow...
i need contact for you
thanks
As of testing there is about a 1-1.5 second delay from talking to hearing it out of speakers.
Use RTSP for session management
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