SIP is the language most business phones speak today. Every IP phone on a desk, every softphone on a laptop and almost every SIP trunk from an Indian operator uses it to set up and end calls. If you are buying an IP PBX, moving off PRI or just trying to understand why a phone shows “not registered”, it helps to know what SIP does and what it does not do. This guide explains what SIP is, how a SIP call works, the ports and security options involved, and where SIP fits in an office phone system.
What is SIP?
SIP stands for Session Initiation Protocol. It is an open standard for starting, changing and ending calls over an IP network. SIP does not carry the voice itself. Its job is the conversation about the call: who is calling whom, whether the other side is free, which audio formats both sides can use, and when the call ends.
Think of SIP as the person at the reception desk who connects two people and then steps away. Once the call is set up, the voice travels separately.
SIP vs VoIP: what is the difference?
People often use the two words as if they mean the same thing. They do not.
| Term | What it means |
|---|---|
| VoIP | Voice over IP: any way of carrying voice over an internet or office data network |
| SIP | One protocol used to set up and manage VoIP calls |
| RTP | The protocol that carries the actual voice packets once the call is set up |
| SDP | A short description inside SIP messages that lists the audio codecs and ports each side offers |
So VoIP is the broad idea, and SIP is the most common way to run it. WhatsApp calls and many apps also use VoIP, but they use their own signalling instead of SIP.
How a SIP call works
Registration
When an IP phone starts, it sends a REGISTER message to the phone system with its extension number and password. The IP PBX checks the credentials and records where that phone can be reached. From then on, calls for that extension go to that device. If registration fails, the phone shows “not registered” and cannot make or receive calls, which is the first thing to check when a phone goes silent.
Setting up a call
- The caller’s phone sends an INVITE to the IP PBX, with an SDP section listing the codecs it supports, such as G.722, G.711 or G.729.
- The PBX finds the destination and forwards the INVITE to the called phone or to a SIP trunk.
- The called phone answers with 180 Ringing while it rings, then 200 OK when the person picks up, along with its own SDP.
- The caller confirms with ACK, and voice starts flowing as RTP packets using the codec both sides agreed on.
- Either side sends BYE to hang up.
Other messages handle transfers, hold and call forwarding, but these five steps cover most of what happens on a normal call.
SIP extensions and SIP trunks
SIP is used in two places in a business phone system.
- SIP extensions connect phones to your PBX. Each desk phone, softphone or mobile app registers as an extension.
- SIP trunks connect your PBX to a telephone operator, so you can call mobiles and landlines and receive calls on your DID numbers. See SIP trunk in India for how operators provide them.
Both use the same protocol, but they are set up differently. An extension usually registers to your PBX. A trunk usually registers from your PBX to the operator, or the operator allows your PBX by its fixed IP address.
In Voxdial IP PBX, each extension has its own password, a limit on how many devices may register to it at the same time, and settings for transport and media encryption. Trunks can be filled in from provider templates for common Indian operators, with a register string supplied by the operator.
SIP ports and transports
| Transport | Default port | Use |
|---|---|---|
| UDP | 5060 | The most common setting for phones and trunks inside India |
| TCP | 5060 | Useful when SIP messages are large or for some NAT setups |
| TLS | 5061 | Encrypts SIP signalling, recommended for phones outside the office |
| WebSocket (WS / WSS) | Set by the PBX | Lets browser-based softphones use SIP |
Voice uses a separate range of RTP ports, set on the PBX. If calls connect but there is no audio, or audio in only one direction, the RTP port range is usually blocked on a firewall or NAT is not set up correctly. Voxdial’s General SIP settings hold the public IP address, local networks and RTP port range in one place for this reason.
SIP security basics
SIP on the open internet attracts scanners looking for weak passwords to make costly international calls. A few rules cut the risk sharply:
- Use long, random passwords on every extension.
- Do not expose port 5060 to the internet unless you need remote phones; use a VPN or limit it by IP address where you can.
- Use TLS for signalling and SRTP for media on phones that work from outside the office, so calls cannot be read on the way.
- Block international dialling for extensions that do not need it.
Voxdial supports UDP, TCP, TLS, WS and WSS transports and has a certificate manager for encrypted connections. For more, read PBX security and toll fraud.
Where SIP fits in an IP PBX
A modern office phone system is mostly SIP from end to end. BitVoice IP phones and softphones register as SIP extensions. The IP PBX handles routing, IVR, queues and recording. SIP trunks bring in outside lines. Older pieces join through gateways: analog phones through an FXS gateway and existing PRI lines through a PRI card or gateway. See FXS vs FXO and what is a PRI line for those parts.
Because SIP is an open standard, you are not tied to one phone brand. Any standards-based SIP phone can register to the PBX, and setup steps are similar across models. See SIP phone configuration for a walkthrough.
Planning a SIP-based phone system for your office? Talk to BitVoice.
FAQs
What is SIP in simple words?
SIP is the protocol phones and phone systems use to start, manage and end calls over an IP network. It sets up the call; the voice itself travels separately as RTP.
Is SIP the same as VoIP?
No. VoIP is any voice call carried over an IP network. SIP is the most common protocol used to set up those calls on business phone systems.
Which port does SIP use?
By default SIP uses port 5060 over UDP or TCP and port 5061 over TLS. Voice uses a separate RTP port range that must also be open on the firewall.
What is the difference between a SIP extension and a SIP trunk?
A SIP extension connects a phone to your PBX. A SIP trunk connects your PBX to a telephone operator so you can make and receive outside calls.
