slash replaced with "or"

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

@@ -44,9 +44,9 @@ Systems that rely solely on peer-to-peer media communication between web browser
 | **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/video media insertion on new or existing media streams. | ✓ | ✓ | ✓ |
+| **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/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. | | ✓ | ✓ |
+| **53.2.6** | [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. | | ✓ | ✓ |
 
 ## V53.3 Signalling