Voice and video communications carry some of an organization’s most sensitive conversations, but the systems that support them are often treated as ordinary network traffic, creating security gaps that attackers may be able to exploit.
Behind the scenes, signaling messages set up and control the session, while separate media streams carry the actual voice or video. If either of these components is left unprotected, attackers may be able to intercept call details, manipulate sessions, or capture media packets that were never meant to be exposed.
That is why securing real-time communications requires attention to both parts of the call path. SIP over Transport Layer Security (TLS) helps protect the signaling messages used to establish and manage the session, while Secure RTP (SRTP) helps protect the media stream that carries the actual conversation. Together, these two technologies provide an important foundation for securing VoIP, UC, and other real-time collaboration services.
In this article, we take a closer look at how SIP over Transport Layer Security (TLS) and Secure RTP (SRTP) work together to protect real-time communications. We’ll examine the difference between signaling and media traffic, what each technology helps secure, and why both are important for protecting VoIP, UC, and collaboration environments.



