SIP trunking is a way to connect a phone system (an IP PBX, a contact center platform or a softswitch) to the public telephone network over the internet or a private IP link, using the SIP protocol instead of physical phone lines. One SIP trunk can carry many simultaneous calls, and capacity is set in channels rather than in copper lines.
What is a SIP trunk?
A SIP trunk is a logical connection between your voice equipment and a provider's network. Nothing physical is installed: the trunk is a set of settings on both sides (IP addresses or credentials, allowed number formats, codecs, channel limits). Calls are set up with SIP (Session Initiation Protocol, defined in RFC 3261) and the audio travels as RTP packets.
The word "trunk" comes from the old telephone network, where trunks were the bundles of lines between exchanges. A SIP trunk does the same job in software: it carries many calls between two systems at once.
How does SIP trunking work?
When a user dials a number, the PBX sends a SIP INVITE to the provider over the trunk. The provider routes the call to its destination, either onto its own network or to another carrier, and returns responses such as 180 Ringing, 200 OK on answer, or an error code if the call fails. Once the call is answered, audio flows in both directions as RTP, encoded with a codec such as G.711 or G.729.
- Outbound calls: your system sends calls to the provider, which terminates them on fixed and mobile networks. In wholesale this side is called VoIP termination.
- Inbound calls: the provider sends calls to your phone numbers (DIDs) over the same trunk. See what a DID number is.
- Authentication: the provider accepts your traffic either by IP address (common between carriers) or by SIP registration with a username and password. Our guide on SIP peering vs registration compares the two.
When a call fails, the SIP response tells you why. Keep the SIP error codes list at hand when you test a new trunk.
What do you need to set up a SIP trunk?
| Item | What to check |
|---|---|
| SIP-capable system | An IP PBX, a dialer, a softswitch or a cloud platform that can send and receive SIP |
| Connectivity | A stable internet or private link with enough bandwidth for your peak calls |
| Channels and CPS | How many concurrent calls and how many new calls per second you need (see SIP trunk channels and CPS) |
| Codecs | G.711 for quality, G.729 to save bandwidth (see G.711 vs G.729) |
| Number format | Usually E.164, sometimes with a technical prefix agreed with the provider |
| Security | A firewall or session border controller in front of the PBX |
As a rule of thumb, one G.711 call uses roughly 80 to 90 kbps in each direction including IP overhead, and one G.729 call around 30 kbps. Multiply by your peak channels and leave headroom.
Is a SIP trunk the same as VoIP?
No. VoIP is the general term for voice over IP networks. A SIP trunk is one specific VoIP service: a connection that links a whole phone system to the telephone network. All SIP trunks are VoIP, but not all VoIP is a SIP trunk. The full comparison is in SIP trunk vs VoIP.
What are the benefits of SIP trunking?
- Capacity on demand: you add channels in settings instead of ordering new lines. A legacy PRI comes in fixed blocks of 23 or 30 channels (see SIP trunk vs PRI).
- Lower cost: you pay for the minutes or channels you use, and international calls can be routed over wholesale carriers.
- Several providers at once: you can connect more than one trunk and route each destination to the best one, which is the basis of least cost routing.
- Numbers from many countries: DIDs from different countries can point to the same system.
The main risks are call quality on a poor internet link and fraud if the PBX is exposed. Monitor jitter and packet loss, restrict destinations you never call, and set credit limits.
Getting SIP trunk routes for your traffic
Businesses with an existing PBX or a hosted PBX usually buy a retail SIP trunk. Call centers, CPaaS platforms and resellers often need wholesale routes per destination, with a clear CLI type, CPS limit and billing increment. VoIPMatcher sources those routes from its carriers and manages them for you, and your data is never shared or sold. See our VoIP termination service, then request a SIP trunk route with your destinations and volumes.
Frequently asked questions
What does SIP stand for?
SIP stands for Session Initiation Protocol. It is the signaling protocol, standardised by the IETF in RFC 3261, that sets up, modifies and ends voice and video calls over IP networks. SIP only handles signaling: the audio itself travels separately as RTP packets between the two endpoints or their media servers.
How many calls can one SIP trunk handle?
There is no fixed number. A SIP trunk carries as many concurrent calls as the channels agreed with the provider and as your bandwidth allows. Small offices often start with 2 to 10 channels, while call centers and carriers run hundreds or thousands, usually with a separate limit on calls per second.
Do I need a PBX to use a SIP trunk?
You need something that speaks SIP on your side: an on-premise IP PBX, a cloud PBX, a contact center platform, a dialer or a softswitch. A single desk phone or softphone normally registers to a hosted service instead of using a trunk. The trunk connects the whole system, not individual handsets.
Can I keep my phone numbers when moving to SIP trunking?
In most countries yes. Geographic and toll-free numbers can usually be ported from the old line provider to the SIP trunk provider through the national porting process. Timelines and documents differ by country and operator, so check with the new provider before you cancel any existing lines.
Is SIP trunking secure?
It can be, if it is configured carefully. Restrict access by IP address, use strong credentials, block destinations you never call, set daily spend limits, and place a firewall or session border controller in front of the PBX. Where the provider supports it, TLS encrypts signaling and SRTP encrypts audio.
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