SIP (Session Initiation Protocol)

Session Initiation Protocol (SIP) is an application-layer protocol that initiates, manages, and terminates real-time communication sessions — including voice calls, video conferences, and instant messaging — across IP networks. SIP is the backbone of modern VoIP and cloud telephony infrastructure.

How SIP Works

SIP handles call signaling (setup and teardown) separately from media transmission (the actual voice data). When a caller dials:

1. SIP INVITE message is sent to the callee's SIP server with details (caller ID, session description).
2. Callee's server rings the device(s) and waits for answer.
3. Upon answer, a SIP 200 OK response is sent back; call is established.
4. Media (voice, video) is transmitted via RTP (separate from SIP signaling).
5. Upon hang-up, BYE message terminates the session.

Advantages of SIP

• Separation of signaling and media: Allows flexible routing and optimization.
• Interoperability: SIP is an open standard; different vendors' systems can interoperate.
• Presence awareness: SIP can convey availability status (online, busy, away).
• Scalability: SIP servers handle thousands of concurrent sessions.

SIP and Cloud Platforms

Cloud CCaaS platforms, softphones, and AI voice agents all use SIP for call setup. This enables seamless integration: an AI voice agent can transfer a call to a softphone-using remote agent, all using SIP signaling.

Security: TLS and SRTP

Standard SIP sends signaling in the clear. For security, SIP over TLS encrypts signaling; SRTP encrypts media. Healthcare and financial institutions require both.