I try to define QoS rules for Telegram’s voice chat feature. My goal is to be able to have downloads running on any of the machines on my network and use the voice chat without having sound issues due to delayed/lost ip packets.
Has anyone here tried that?
It would be great if someone could point me in the right direction.
These are the general principles I would follow for myself if I were to solve this problem:
- read carefully how IPFire QoS works;
- try to find the documentation of the audio chat API for telegram to define protocol (UDP, TCP) and ports range involved so that you can define a class of traffic in QoS; If you can’t find this information, you need to use tools like
wiresharkon your own system to find which protocol and ports you need to define a QoS class;
- if the protocol does not allow you to define precise ports, as I suspect this is the case, either you are SOL, or you could install a reverse proxy on your system and limit there the usable ports so that you can define a specific class;
- when you have a Class to work on, then you need to experiment on priority (high I guess) and size of the pipe you need to reserve. I believe trial and error is the only way.
I hope someone more qualified than I am can give you more useful information than those I just wrote.