In short
SIP vs VoIP is not a choice between two products. VoIP (voice over IP) is the technology that carries phone calls as data over an IP network. SIP (Session Initiation Protocol) is the signalling protocol most VoIP systems use to set up, change and end those calls, while a second protocol, RTP, carries the audio itself.
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VoIP is the technology, SIP is a protocol
VoIP is the broad term for any way of carrying voice over an internet protocol network. It covers the codec that turns speech into data, the packets that cross the network, the phone system that routes calls and the devices people talk into.
SIP is one specific part of that picture. The IETF standard that defines it, RFC 3261, describes SIP as "an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants", including internet telephone calls. In plain terms, SIP does the dialling, ringing, answering, transferring and hanging up.
SIP does not carry the conversation. Once a call is set up, the audio travels as RTP (Real-time Transport Protocol, RFC 3550) packets, and SIP messages carry a session description so both ends agree on a codec. So the relationship between VoIP and SIP is simple: SIP is the call-control language that most VoIP phone systems, phones and trunks speak. For the full journey of a call, read how a VoIP phone system works.
SIP vs VoIP at a glance
| VoIP | SIP | |
|---|---|---|
| What it is | A technology: voice carried over IP networks | A signalling protocol, defined by the IETF in RFC 3261 |
| What it does | Everything needed to make a call over a data network | Sets up, changes and ends calls; handles registration of phones |
| Carries the audio? | Yes, as data packets | No; the audio travels as RTP |
| Alternatives | Traditional analogue and ISDN telephony | Older standards such as H.323, and platforms' own protocols |
| Where you meet it | Phone systems, apps, desk phones, digital landlines | SIP phones, SIP trunks, SIP settings on routers and firewalls |
| Something you buy? | Yes, as a VoIP phone system or service | Not on its own; it comes inside phones, phone systems and trunks |
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How SIP sets up a VoIP call
A simplified SIP call between two phones on the same phone system looks like this:
- Register. Each phone sends a SIP REGISTER message so the phone system knows where to reach it.
- Invite. The caller's phone sends an INVITE, with a session description listing the codecs it can use.
- Ring. The phone system finds the called phone and passes the INVITE on; the caller hears ringing.
- Answer. The called phone answers with a 200 OK, including its own choice of codec, and the caller confirms with an ACK.
- Talk. Audio flows as RTP packets in both directions, either directly between the phones or through the phone system; SIP stays quiet in the background.
- Hang up. Either side sends a BYE and the call ends.
By default, SIP uses port 5060, or 5061 when the signalling is encrypted with TLS. Calls to ordinary phone numbers follow the same pattern, with the phone system handing the call to the public network. We explain those routes in how a VoIP phone system works.
Is SIP the same as VoIP? Where SIP fits on each platform
Not quite. Most VoIP uses SIP somewhere, but some platforms use their own protocols internally and speak SIP at the edges. The platforms we supply show the range:
- 3CX is built around SIP. 3CX lists supported IP phones from Fanvil, Snom and Yealink, plus DECT systems, door phones and adapters, and supported SIP trunks for the connection to the phone network. See 3CX VoIP.
- Microsoft Teams Phone apps and certified devices sign in to Teams itself. SIP appears at the edges: Direct Routing connects a Microsoft-certified Session Border Controller to Teams Phone over SIP, and SIP Gateway lets compatible SIP phones sign in to Teams. See Teams VoIP.
- Webex Calling runs in Cisco's cloud. Where you use your own carrier or PBX, Cisco documents a premises-based local gateway that connects to Webex Calling over a SIP trunk, either registration-based or certificate-based. See Webex VoIP.
- Forever UC and Forever Teams handle the SIP side for you: Forever UC is app-first, and Forever Teams is built on Direct Routing.
For you as a buyer, this matters mainly when choosing phones, connecting an existing phone system or setting up a firewall.
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SIP phones vs VoIP phones
"VoIP phone", "IP phone" and "SIP phone" are often used interchangeably, but they are not quite the same:
- VoIP phone or IP phone: any phone that makes calls over an IP network, whatever protocol it uses.
- SIP phone: an IP phone that speaks standard SIP, so it can register to any SIP-based phone system that supports it.
- Platform phones: phones that run a platform's own software, such as Teams certified phones, which sign in to Teams directly.
Some phones can do both. Microsoft lists Cisco IP phones with multiplatform SIP firmware, and SIP devices from vendors such as Poly, Yealink and AudioCodes, as compatible with Teams through SIP Gateway. Before you reuse existing handsets, check that the platform supports them and can provision them automatically. Our guide to VoIP handsets for business covers compatibility by platform, and softphone vs desk phone helps you decide who needs a handset at all.
SIP trunks: one way to connect a phone system
A SIP trunk uses SIP to connect a phone system you run, on site or in your own cloud account, to the public phone network over an internet connection. It is how an IP PBX replaces its ISDN or analogue lines, and how Teams Direct Routing connects to a carrier.
With a hosted VoIP phone system you usually do not need to think about trunks, because the connection to the phone network comes with the service. If you are keeping your own PBX, compare SIP trunk providers on our sister site, which also explains how many SIP channels you need. Moving off ISDN? See ISDN to VoIP, and for how a PBX relates to VoIP, PBX vs VoIP.
SIP settings and security
Because SIP controls who can make calls, it deserves care:
- SIP ALG. Many routers have a SIP application layer gateway that rewrites SIP messages and often causes one-way audio or registration failures. 3CX advises disabling it, and we can check the guidance for whichever platform you choose. See VoIP router.
- Encryption. The NCSC recommends TLS for signalling and SRTP for media streams on cloud and hosted PBX systems, protecting calls against eavesdropping.
- Lock it down. The NCSC advises restricting a PBX's inbound and outbound traffic to trusted IP addresses, such as your SIP trunk provider, and changing every default password.
- Limit dialling. Disable international and premium-rate calling where you do not need it, to reduce the risk of dial-through fraud.
Our VoIP security guide covers the risks and controls in full.
How we help
You do not need to become a SIP expert to buy a good phone system. Forever Systems supplies hosted 3CX, Microsoft Teams Phone, Webex Calling, Forever UC, Forever Teams and SIP trunks, and we can handle the SIP detail for you: compatible phones, trunks, firewall and router settings, and encryption where the platform supports it.
Contact us and tell us what you have today, whether that is a PBX, ISDN lines or a few analogue phones, and we can set out the simplest route to VoIP.
How Forever Systems helps you choose a VoIP phone system
We supply and support several VoIP platforms, so we can recommend the one that fits how your team takes calls rather than fitting your business around one product.
- Platform advice across hosted 3CX, Microsoft Teams Phone, Webex Calling, Forever UC and Forever Teams, with SIP trunks for phone systems you already own
- A requirements check of your broadband, network and routers before anything goes live
- Numbers and porting: we can help you keep your existing business numbers and plan the move away from analogue and ISDN lines
- Handsets and headsets that suit each platform, bought outright or, in some cases, included in your monthly rental, with the options confirmed at quote
- Set-up and changes: call flows, users and devices configured for you, and help with moves, adds and changes afterwards
- For BT Cloud Voice, see our sister site cloudvoice.cloud, and for hosted and cloud platforms in depth, cloudphone.cloud
We work with businesses, schools, charities and public sector organisations.
Frequently asked questions
Is SIP the same as VoIP?
No. VoIP is the technology that carries calls over IP networks. SIP is the signalling protocol most VoIP systems use to set up, change and end calls. The audio itself travels separately, as RTP packets.
What is the relationship between VoIP and SIP?
SIP is a building block of VoIP. It handles call control, such as registering phones, ringing, answering, transferring and hanging up, while other parts of VoIP encode the voice and carry it across the network.
What is SIP in VoIP?
SIP (Session Initiation Protocol) is the IETF standard, RFC 3261, that VoIP phones, phone systems and trunks use to start and end calls. By default it uses port 5060, or 5061 when encrypted with TLS.
What is the difference between a SIP phone and a VoIP phone?
Every SIP phone is a VoIP phone, but not every VoIP phone uses SIP. A SIP phone speaks the open SIP standard, so it can register to SIP-based phone systems. Some VoIP phones run a platform's own software instead, such as Teams certified phones.
Do I need SIP trunks for VoIP?
Only if you run your own phone system, such as an on-site or self-hosted IP PBX, or connect Teams through Direct Routing. With a hosted VoIP phone system, the connection to the phone network is part of the service.
Should SIP ALG be on or off?
In most cases off. SIP ALG on routers often rewrites SIP messages and causes one-way audio or phones that will not register. 3CX advises disabling it; check your platform's guidance before changing router settings.
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