What is SIP trunking?
SIP trunking connects your business phone system to the public telephone network over an internet connection — replacing physical ISDN and PRI lines. This guide covers how it works, how it compares to PRI, what you need, and when to use it.
What Is SIP Trunking?
SIP trunking is a method of delivering telephone service over an internet connection using the Session Initiation Protocol (SIP). It replaces traditional PSTN connections — ISDN lines, PRI circuits — with a virtual link between your business phone system and a SIP trunk provider.
Where a physical phone line uses copper or fibre circuits dedicated to voice, a SIP trunk carries voice calls as data packets over your existing IP network. The SIP trunk provider handles the interconnection to the public switched telephone network (PSTN) on your behalf. From the outside, calls appear no different — the technology that carries them is fundamentally changed.
SIP trunking is the carrier connectivity layer. It sits between your business phone system or IP-PBX and the wider telephone network. It is distinct from the PBX itself, which handles the internal routing, extensions, and features.
How SIP Trunking Works
Understanding SIP trunking is easier when you follow the path of a call from start to finish.
- Your IP-PBX or UCaaS platform. This is the switching and routing layer for your business — managing extensions, ring groups, auto-attendant menus, and call queues. It may be on-premise hardware or a cloud-hosted service. It communicates using SIP.
- The SIP trunk. Your PBX registers to your SIP trunk provider using credentials over your internet connection. The provider allocates a number of channels — each channel is the capacity for one concurrent call. If you have twenty channels, twenty calls can be active simultaneously.
- The SIP trunk provider. The provider operates the infrastructure that bridges your SIP connection to the PSTN. When someone outside your business calls one of your DID (Direct Inward Dial) numbers, the call arrives at the provider's platform and is delivered to your PBX over the SIP trunk.
- The PSTN or destination VoIP network. Outbound calls from your PBX travel through the provider's platform and are routed to their destination — a mobile, a landline, or another VoIP network — via standard telephone network interconnections.
Authentication between your PBX and the SIP provider is handled either by IP address (your PBX's public IP is registered as authorised) or by username and password credentials. DID numbers assigned to your account are delivered to your PBX when called, and your PBX presents those numbers as caller ID on outbound calls.
SIP Trunking vs PRI
Before SIP trunking became standard, businesses connected their PBX to the PSTN using PRI (Primary Rate Interface) — a physical T1 or E1 digital circuit. The comparison matters because many businesses still run PRI and are evaluating whether to migrate.
| PRI (T1/E1) | SIP Trunk | |
|---|---|---|
| Delivery | Physical T1 or E1 circuit to your premises | Internet connection (broadband, MPLS, or dedicated IP) |
| Capacity | Fixed: 23 channels (T1) or 30 channels (E1) | Elastic: add or remove channels on demand |
| Contract | Typically long-term with hardware commitments | Month-to-month options common; no physical line contract |
| Hardware | Carrier-managed demarcation equipment at your site | No carrier hardware on-site; managed via portal |
| Cost structure | Fixed monthly line rental regardless of usage | Pay per channel; some providers offer metered call rates |
| Geographic flexibility | Number tied to physical circuit location | DID numbers in any area code regardless of office location |
For businesses with existing PRI circuits, the migration path to SIP trunking is well established. The PBX hardware (if SIP-capable) typically remains in place; only the carrier connectivity changes.
Benefits of SIP Trunking
- No physical lines. There is no copper or fibre circuit to provision, maintain, or repair at the carrier level. If your internet connection is working, your telephony is working.
- Flexible capacity. Channels can be scaled up or down as your call volumes change — during seasonal peaks, office expansions, or reductions. You are not locked to the fixed 23-channel increment of a PRI circuit.
- Geographic number flexibility. You can obtain DID numbers with any area code without having a physical presence in that location. A business headquartered in one city can present local numbers to customers in multiple regions.
- Failover and redundancy. SIP trunks can be configured to failover to a secondary provider, a different IP address, or mobile numbers if your primary internet connection fails. This is significantly easier to arrange than PRI failover, which requires physical circuit redundancy.
- Lower per-channel cost. Eliminating physical circuit infrastructure reduces the carrier's cost base, and that is reflected in per-channel pricing. Businesses with high call volumes often see meaningful cost reductions compared with PRI line rental.
What You Need for SIP Trunking
Three things are required before you can use SIP trunking:
- A compatible IP-PBX or UCaaS platform. Your phone system must support SIP as a trunk protocol. Most modern IP-PBX systems — both on-premise and cloud-hosted — do. If you are running a legacy analogue or digital PBX, you will need a media gateway or ATA to bridge the signalling before you can connect a SIP trunk.
- A stable internet connection with sufficient bandwidth. Each concurrent call consumes approximately 85 kbps with the G.711 codec (uncompressed) or approximately 32 kbps with G.729 (compressed). A business expecting up to twenty simultaneous calls on G.711 should allocate at least 1.7 Mbps dedicated to voice. Quality of Service (QoS) configuration on your router ensures voice packets are prioritised over general internet traffic.
- A SIP trunk provider account. You register with a provider, choose your channel count, obtain DID numbers, and receive the SIP server credentials to enter into your PBX configuration. Providers vary in geographic number availability, pricing models, and support for specific PBX platforms.
If you are evaluating wholesale voice connectivity for a larger deployment or carrier use case, the requirements extend to include IP peering agreements and traffic volume commitments.
SIP Trunking Use Cases
- Multi-site businesses. A single SIP trunk account can serve multiple office locations. Each site connects its PBX to the same provider, sharing a pool of channels and presenting consistent numbers regardless of which site a call originates from.
- Contact centres. High inbound and outbound call volumes benefit from elastic channel capacity. SIP trunking allows a contact centre to scale channels up during peak periods and reduce them during quieter periods — without physical line provisioning delays. Pairing with an IVR for inbound routing is a common configuration.
- Businesses replacing legacy PRI or ISDN lines. As carriers wind down ISDN and PRI services in various markets, SIP trunking is the standard replacement path. The PBX typically stays in place; only the carrier connection changes.
- Wholesale carriers and resellers. Carriers buying interconnect minutes in volume use SIP trunking as the delivery mechanism. See SIP trunking vs PRI for a deeper comparison relevant to carrier-side decisions.
Frequently Asked Questions
What is a SIP trunk?
A SIP trunk is a virtual telephone line delivered over an internet connection using the Session Initiation Protocol (SIP). It connects a business IP-PBX or UCaaS platform to a SIP trunk provider, which in turn connects to the public switched telephone network (PSTN). Unlike a physical phone line, a SIP trunk has no hardware at the carrier end — capacity is defined by the number of channels (concurrent calls) you configure with your provider.
How many calls can a SIP trunk handle at once?
A SIP trunk can handle as many concurrent calls as the number of channels you have provisioned. Each active call occupies one channel. If you have ten channels and ten calls are in progress simultaneously, an eleventh caller will receive a busy signal unless you add more channels. Most SIP trunk providers allow you to adjust channel counts on demand, which is one of the key advantages over physical PRI circuits where capacity is fixed by the hardware.
What is the difference between SIP trunking and VoIP?
VoIP (Voice over Internet Protocol) is the broad technology category that describes carrying voice calls as data packets over IP networks. SIP trunking is a specific application of VoIP — it is the method used to connect a business phone system to the public telephone network over an internet connection using SIP as the signalling protocol. All SIP trunking is VoIP, but not all VoIP involves SIP trunking. For example, a hosted cloud PBX may use VoIP internally while the carrier connectivity to the PSTN is handled through SIP trunks managed by the provider.
Do I need special hardware for SIP trunking?
You need a SIP-compatible IP-PBX or UCaaS platform that can register to a SIP trunk provider. Many modern IP-PBX systems — both on-premise and cloud-hosted — support SIP trunking natively. If you have an older PBX that uses digital or analogue interfaces, you may need an Analog Telephone Adapter (ATA) or a media gateway to convert the signalling. Beyond the PBX itself, you need a stable internet connection with sufficient bandwidth — no additional hardware is required at the carrier side.
What bandwidth do I need for SIP trunking?
The required bandwidth depends on the codec used and the number of concurrent calls. With the G.711 codec (uncompressed audio), each concurrent call requires approximately 85 kbps of bandwidth when accounting for headers and overhead. With compressed codecs such as G.729, that figure drops to around 32 kbps per call. As an example, a business expecting up to ten simultaneous calls should provision at least 850 kbps dedicated to voice when using G.711. Quality of Service (QoS) configuration on your router is also important — it ensures voice traffic is prioritised over other internet usage during periods of contention.
Is SIP trunking the same as a hosted PBX?
No. SIP trunking and hosted PBX are related but distinct services. A hosted (cloud) PBX is a complete phone system where the switching, routing, and feature logic are managed by a provider. SIP trunking is specifically the carrier connectivity layer — it provides the channel between your phone system and the PSTN. A hosted PBX provider typically uses SIP trunking internally to connect their platform to the telephone network, but that is managed on your behalf. When you buy SIP trunks directly, you are responsible for managing your own PBX and connecting it to the trunks yourself.
SIP trunking and wholesale voice connectivity
MCC provides SIP trunking, wholesale voice, and cloud communications infrastructure for businesses and carriers. Contact us to discuss your connectivity requirements.