Glossary · SIP
What is SIP?
SIP (Session Initiation Protocol) is the signaling standard that establishes, manages, and ends real-time communication sessions over IP networks — most commonly VoIP voice and video calls. SIP is the “dial tone” of internet telephony: it handles setting up a call, locating the other party, negotiating how the media will flow, and tearing the call down when it ends. It does not carry the audio itself; it coordinates the session while a separate media protocol transports the voice.
How SIP works
When a SIP call is placed, the protocol exchanges a series of text-based messages — much like HTTP requests — between the endpoints:
- INVITE requests a session with another party.
- Responses (ringing, OK) report progress and acceptance.
- ACK confirms the connection is established.
- BYE ends the session.
During setup, SIP negotiates the media details (codecs, ports) so both sides agree on how the audio or video will be sent. The actual voice then travels over RTP/SRTP, while SIP stays in the background managing the session state.
Common SIP methods and response codes
SIP defines a small set of request “methods” plus numbered response codes that deliberately mirror HTTP. The core methods:
| Method | Purpose |
|---|---|
| INVITE | Request or establish a session |
| ACK | Confirm the session is set up |
| BYE | End an active session |
| REGISTER | Tell the registrar where a user is reachable |
| CANCEL | Abort a pending INVITE |
| OPTIONS | Query a server’s capabilities |
Responses group the same way HTTP does: 1xx provisional (100 Trying, 180 Ringing), 2xx success (200 OK), 3xx redirection, and 4xx–6xx failures such as 486 Busy Here or 404 Not Found.
What a SIP address and SIP ports look like
A SIP address — or SIP URI — identifies an endpoint in the form sip:user@domain, much like an email address (for example sip:alice@dialphone.com). It lets SIP locate the party to call regardless of which physical device they are signed in on. SIP signaling travels on port 5060 for unencrypted UDP or TCP and port 5061 for TLS-encrypted (SIPS) sessions, while the negotiated RTP media uses a separate dynamic port range. These ports matter when writing firewall rules, since a blocked 5060/5061 silently breaks call setup.
SIP vs. SIP trunking vs. VoIP
These terms are tightly related and often confused:
- VoIP is the broad concept of carrying voice over the internet.
- SIP is the specific protocol that signals and controls those calls.
- SIP trunking is the service that uses SIP to connect a business phone system to the carrier network, replacing physical phone lines.
In short: VoIP is the “what,” SIP is the “how,” and SIP trunking is the commercial service built on SIP.
Where SIP is used
- Connecting IP phones, softphones, and PBXs to carriers
- Business SIP trunking replacing legacy lines
- Video conferencing and unified communications signaling
- Microsoft Teams Direct Routing and similar cloud-calling integrations
Because SIP traffic crosses network boundaries, it is typically secured and normalized by a Session Border Controller, which protects the network and resolves differences between SIP implementations.
SIP frequently asked questions
What is the difference between SIP and VoIP?
VoIP is the general technology of making calls over the internet. SIP is the specific signaling protocol that sets up and manages those calls. You can run VoIP with SIP, which is the dominant approach, so the terms are related but not interchangeable — SIP is one component of how VoIP works.
What is the difference between SIP and SIP trunking?
SIP is the protocol; SIP trunking is a service that uses it. SIP trunking delivers voice connectivity to a business phone system over the internet using SIP signaling, replacing traditional physical phone lines and connecting the system to the wider telephone network.
Does SIP carry the actual voice audio?
No. SIP only handles signaling — setting up, modifying, and ending sessions. The voice or video media travels over a separate protocol, RTP (or encrypted SRTP), whose parameters SIP negotiates during call setup.
Is SIP secure?
SIP can be secured with TLS for the signaling and SRTP for the media, and is typically protected by a Session Border Controller at the network edge to prevent fraud and eavesdropping. Unsecured SIP exposed directly to the internet is a common target for toll fraud, so encryption and an SBC are standard for business use.
See how DialPhone uses SIP
DialPhone runs on SIP-based infrastructure with SIP trunking and session border control managed for you, so businesses get secure, carrier-grade cloud calling without configuring SIP themselves.