SIP (Session Initiation Protocol) is a text-based signaling protocol, modeled on HTTP, that creates, changes and terminates communication sessions such as voice and video calls. It does not carry the audio itself; it only negotiates who is calling whom and which media formats and addresses to use.
Why it matters
SIP is the common language between IP phones, PBX systems and VoIP carriers. Because it is a separate signaling layer, the same protocol can set up calls, conferences and video sessions across vendors. It normally listens on port 5060 (UDP or TCP) and 5061 for TLS-encrypted signaling.
Example
A simplified call setup:
INVITE sip:alice@example.com SIP/2.0
From: <sip:bob@example.net>
To: <sip:alice@example.com>The callee answers with 180 Ringing, then 200 OK, and the caller confirms with ACK. Both sides exchange SDP descriptions inside these messages that state the IP addresses and ports for the RTP media streams.
Common mistakes
- Leaving SIP on port 5060 open to the whole internet, which invites brute-force registration attempts and toll fraud.
- Using NAT without SIP-aware handling, so the media addresses inside the message are private and calls have one-way audio.
- Assuming signaling encryption (TLS) also encrypts the media; that needs SRTP.
Related
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