CPAN Proxy Authentication Needed

The main task is to monitor traffic on the local network, as all requests will be handled by an organized proxy. Most often it is used to block access to certain resources in offices.

Changing the WebRTC (Web Real-Time Communication) value in Selenium involves modifying the browser's configuration options. WebRTC settings are not directly exposed through Selenium WebDriver, so you need to use browser-specific options or preferences.

Below are examples for changing WebRTC settings in Chrome and Firefox using Selenium in Python. Keep in mind that the availability of certain options may vary depending on the browser version, and these examples may need adjustments based on your specific requirements.

from selenium import webdriver
from selenium.webdriver.chrome.options import Options

chrome_options = Options()

# Disable WebRTC
chrome_options.add_argument('--disable-webrtc')

# Other options (customize as needed)
# chrome_options.add_argument('--use-fake-device-for-media-stream')
# chrome_options.add_argument('--use-fake-ui-for-media-stream')

driver = webdriver.Chrome(chrome_options=chrome_options)

# Your Selenium script...

driver.quit()

In this example, --disable-webrtc is used to disable WebRTC. You can explore other Chrome command-line options related to WebRTC here.

Firefox

from selenium import webdriver
from selenium.webdriver.firefox.options import Options

firefox_options = Options()

# Disable WebRTC
firefox_options.set_preference('media.peerconnection.enabled', False)

# Other preferences (customize as needed)
# firefox_options.set_preference('media.navigator.streams.fake', True)
# firefox_options.set_preference('media.navigator.permission.disabled', True)

driver = webdriver.Firefox(firefox_options=firefox_options)

# Your Selenium script...

driver.quit()

In this example, media.peerconnection.enabled is set to False to disable WebRTC in Firefox. Additional preferences can be adjusted based on your needs. You can find more Firefox preferences related to WebRTC here.

Remember that changing browser preferences may have implications on the behavior of your application, and modifying settings like WebRTC should be done responsibly and in accordance with the terms of service of the websites you are interacting with.

The maximum size of an RTP (Real-time Transport Protocol) packet when transmitted over TCP/UDP protocol depends on the payload size and the addition of RTP header information.

RTP is a transport protocol specifically designed for real-time applications like audio and video streaming. It is typically used in conjunction with UDP or TCP, as it does not provide its own transport layer.

RTP packets consist of two parts:

1. Payload: This is the actual data being transmitted, which can be audio, video, or other real-time data. The payload size is determined by the application or codec being used.

2. Header: The RTP header contains metadata required for the proper processing and synchronization of the payload. The header has a fixed size of 12 bytes. The maximum size of an RTP packet can be calculated by adding the payload size and the fixed header size:

Maximum RTP packet size = Payload size + 12 bytes (RTP header)

The payload size depends on the application or codec being used. For example, if you're using an audio codec that generates 100-byte audio frames, the maximum RTP packet size would be:

Maximum RTP packet size = 100 bytes (payload) + 12 bytes (RTP header) = 112 bytes

In the case of video codecs, the payload size can be significantly larger, depending on the video resolution, compression, and frame rate.

When RTP is used over TCP or UDP, the maximum size of the RTP packet is limited by the maximum payload size supported by the underlying transport protocol. For TCP, the maximum segment size (MSS) is determined by the MTU (Maximum Transmission Unit) of the network and the TCP header size. For UDP, the maximum packet size is limited by the MTU of the network and the UDP header size.

In summary, the maximum size of an RTP packet when transmitted over TCP/UDP protocol depends on the payload size and the addition of RTP header information, as well as the underlying transport protocol's limitations.

If you have the operating system Ubuntu, the messenger will not be able to connect to the proxy. On other operating systems, if this situation occurs, you should update your application to the latest version. Another reason for no connection may be a server restart. In this case, you should either wait for the traffic to decrease or connect to a new proxy. Sometimes, to get Telegram working via proxy again, you simply need to replace the outdated proxy server with a new one.

There are HTTP proxy, FTP proxy, SOCKS proxy, SMTP proxy, CGI proxy. They differ only in the data transmission protocol used and the purpose for which they are used. For example, SMTP proxy allows you to organize a secure server for e-mail.