A signaling protocol used to establish, modify, and terminate real-time communication sessions over IP networks. These sessions may include voice calls, video calls, instant messaging, and other multimedia interactions. SIP operates at the application layer and works in conjunction with other protocols—such as RTP (Real-time Transport Protocol)—to deliver the actual media once a session is established.
When security is a concern, SIP is often paired with TLS (Transport Layer Security) to create SIP over TLS. This approach encrypts SIP signaling messages, protecting them from interception or tampering as they travel between endpoints or through intermediary SIP servers. By encrypting the signaling layer, sensitive information such as phone numbers, authentication credentials, and call metadata is kept confidential.
SIP over TLS does not encrypt the media itself; for complete end-to-end confidentiality, Secure RTP (SRTP) is used alongside it to protect the actual audio and video streams. Together, SIP over TLS and SRTP ensure that both the signaling and the media remain secure from eavesdropping and tampering.
In modern VoIP deployments—especially in enterprise and hosted cloud PBX environments—using SIP over TLS is considered a best practice. It not only provides encryption but also helps avoid issues with certain network firewalls and NAT configurations, as TLS traffic often uses TCP port 5061, which is less likely to be blocked than the standard SIP UDP port 5060.
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