Refactor certain requirements

5.0/en/0x53-V53-WebRTC.md

@@ -35,30 +35,32 @@ These requirements only apply to systems that host TURN servers as part of their
 
 Media security is paramount in WebRTC applications, as it directly impacts the confidentiality, integrity and availability of audio and video communications. By addressing these security concerns, developers can safeguard the media streams in WebRTC applications, preventing eavesdropping, tampering, and denial-of-service attacks that could compromise user privacy and communication quality.
 
-These requirements only apply to systems that host their own WebRTC media servers, such as Selective Forwarding Units (SFUs), Multipoint Control Units (MCUs), recording servers, or gateway servers. These servers are responsible for handling, routing, processing, or storing media streams within the application. The security of these media servers is crucial because they manage and distribute media between peers, and improper security could lead to unauthorized access, interception, or manipulation of media streams.
+These requirements only apply to systems that host their own WebRTC media servers, such as Selective Forwarding Units (SFUs), Multipoint Control Units (MCUs), recording servers, or gateway servers. These servers are responsible for handling, routing, processing, or storing media streams within the application. The security of these media servers is crucial because they manage and distribute media between peers, and improper security could lead to unauthorized access, interception, or manipulation of media streams. 
 
-Systems that rely solely on peer-to-peer media communication between web browsers, without the involvement of intermediate media servers, are excluded from these specific media-related security requirements. However, such systems should still adhere to the other requirements outlined in this chapter to ensure the overall security of their communications.
+In particular, it will be necessary to implement protections against flood attacks such as rate limiting, validating timestamps, using synchronized clocks to match real-time intervals, and managing buffers to prevent overflow and maintain proper timing. If packets for a particular media session arrive too quickly, excess packets should be dropped. It is also important to protect the system from malformed packets by implementing input validation, safely handling integer overflows, preventing buffer overflows, and employing other robust error-handling techniques.
+
+Systems that rely solely on peer-to-peer media communication between web browsers, without the involvement of intermediate media servers, are excluded from these specific media-related security requirements. However, such systems should still adhere to the other requirements outlined in this chapter to ensure the overall security of their communications. 
 
 | # | Description | L1 | L2 | L3 |
 | :---: | :--- | :---: | :---: | :---: |
 | **53.2.1** | [ADDED] Verify that the key for the Datagram Transport Layer Security (DTLS) certificate is private by ensuring it is not reused in existing products or open-source projects and confirming it is not distributed or leaked. | ✓ | ✓ | ✓ |
 | **53.2.2** | [ADDED] Verify that the media server is configured to use and support strong cipher suites for the Datagram Transport Layer Security (DTLS) exchange, ensuring that the selected cipher suites are considered strong and secure. | ✓ | ✓ | ✓ |
 | **53.2.3** | [ADDED] Verify that the media server is not susceptible to the "WebRTC DTLS ClientHello Race Condition" vulnerability by checking if the media server is publicly known to be vulnerable or by performing the race condition test. | | ✓ | ✓ |
-| **53.2.4** | [ADDED] Verify that Secure Real-time Transport Protocol (SRTP) authentication is checked at the media server to prevent Real-time Transport Protocol (RTP) injection attacks from causing DoS or audio or video media insertion on new or existing media streams. | ✓ | ✓ | ✓ |
-| **53.2.5** | [ADDED] Verify that the media server is able to continue processing incoming media traffic during a flood of Secure Real-time Transport Protocol (SRTP) packets from legitimate users. This should be achieved by implementing rate limiting, validating timestamps, using synchronized clocks to match real-time intervals, and managing buffers to prevent overflow and maintain proper timing. If packets for a particular media session arrive too quickly, excess packets should be dropped. | | ✓ | ✓ |
-| **53.2.6** | [ADDED] Verify that any audio or video recording mechanisms associated with the media server are able to continue processing incoming media traffic during a flood of Secure Real-time Transport Protocol (SRTP) packets from legitimate users by implementing rate limiting, validating timestamps, using synchronized clocks to match real-time intervals, and managing buffers to prevent overflow and maintain proper timing. If packets for a particular media session arrive too quickly, excess packets should be dropped. | | ✓ | ✓ |
-| **53.2.7** | [ADDED] Verify that the media server is able to continue processing incoming media traffic when encountering malformed SRTP packets. This should be achieved by implementing input validation, safely handling integer overflows, preventing buffer overflows, and employing other robust error-handling techniques. | | ✓ | ✓ |
+| **53.2.4** | [ADDED] Verify that Secure Real-time Transport Protocol (SRTP) authentication is checked at the media server to prevent Real-time Transport Protocol (RTP) injection attacks from leading to either a Denial of Service condition or audio or video media insertion into media streams. | ✓ | ✓ | ✓ |
+| **53.2.5** | [ADDED] Verify that the media server is able to continue processing incoming media traffic during a flood of Secure Real-time Transport Protocol (SRTP) packets from legitimate users. | | ✓ | ✓ |
+| **53.2.6** | [ADDED] Verify that any audio or video recording mechanisms associated with the media server are able to continue processing incoming media traffic during a flood of Secure Real-time Transport Protocol (SRTP) packets from legitimate users. | | ✓ | ✓ |
+| **53.2.7** | [ADDED] Verify that the media server is able to continue processing incoming media traffic when encountering malformed SRTP packets. | | ✓ | ✓ |
 
 ## V53.3 Signalling
 
-Signalling is a critical component of WebRTC applications, responsible for coordinating communication sessions between peers. The security of the signalling process is essential to prevent unauthorized access, eavesdropping, and service disruptions.
+Signalling is a critical component of WebRTC applications, responsible for coordinating communication sessions between peers. The security of the signalling process is essential to prevent unauthorized access, eavesdropping, and service disruptions. This protecting the system fralfo by implementing input validation, safely handling integer overflows, preventing buffer overflows, and employing other robust error-handling techniques.
 
 These requirements only apply to systems that host signalling servers as part of their WebRTC infrastructure.
 
 | # | Description | L1 | L2 | L3 |
 | :---: | :--- | :---: | :---: | :---: |
 | **53.3.1** | [ADDED] Verify that the signalling server is able to continue processing incoming signalling messages during a flood attack. This should be achieved by implementing rate limiting at the signalling level. | | ✓ | ✓ |
-| **53.3.2** | [ADDED] Verify that the signalling server is able to is able to continue processing signalling messages when encountering malformed signalling messages. This should be achieved by implementing input validation, safely handling integer overflows, preventing buffer overflows, and employing other robust error-handling techniques. | | ✓ | ✓ |
+| **53.3.2** | [ADDED] Verify that the signalling server is able to is able to continue processing signalling messages when encountering malformed signalling messages. | | ✓ | ✓ |
 
 ## References