SIP Trunking Explained: Moving Business Phones to the Cloud
SIP trunking replaces traditional PSTN and ISDN telephone lines with an internet-based voice service, letting businesses make and receive calls over their existing data connection. It reduces line rental costs, simplifies number management, and scales instantly without waiting for a carrier to provision physical circuits. This guide explains how SIP trunking works, what you need to get started, and how it compares to hosted PBX and UCaaS.
What Is SIP?
SIP (Session Initiation Protocol) is a signalling protocol used to set up, manage, and tear down real-time communication sessions — primarily voice and video calls — over IP networks. SIP handles the call setup (who is calling whom, codec negotiation, ringing) while the actual audio travels as a separate media stream using the RTP (Real-time Transport Protocol). SIP is an open standard defined by the IETF, making it vendor-neutral and widely supported by IP phones, PBX systems, and carriers around the world.
What Does a SIP Trunk Replace?
Traditionally, businesses connected to the public telephone network through PSTN analogue lines or ISDN (PRI/BRI) digital circuits. Each line or channel represented one concurrent call path and was delivered over dedicated copper or fibre from the carrier. Adding more lines meant ordering physical circuits, waiting for provisioning, and paying monthly line rental for each.
A SIP trunk replaces all of those physical lines with a virtual connection over your internet service. Instead of 10 ISDN channels, you purchase a SIP trunk with 10 concurrent call paths — or 20, or 50 — and they are delivered instantly over your existing data link. There is no physical circuit to install, and scaling up or down is a configuration change rather than a truck roll.
How SIP Trunking Works
Your on-premises PBX (such as a 3CX, FreePBX, Yealink, or Cisco system) connects to a SIP trunk provider over the internet or a dedicated data link. When an internal user dials an external number, the PBX sends a SIP INVITE message to the provider's Session Border Controller (SBC). The provider routes the call to the public telephone network (PSTN) and the call connects. Inbound calls work in reverse — the provider receives the call from the PSTN and delivers it to your PBX via SIP.
DID numbers (Direct Inward Dialling) are mapped to the SIP trunk, so each employee or department can have their own direct number without needing a separate physical line. You can port existing numbers from your current carrier or provision new numbers from the SIP provider, often within minutes.
Benefits of SIP Trunking
The advantages of SIP trunking over traditional telephony are substantial:
- Cost savings: SIP call rates are typically 30–60% lower than PSTN rates, and there is no per-line rental. International and long-distance calls see the biggest savings.
- Flexibility: Add or remove concurrent call channels on demand. No minimum contract for channels in many cases.
- Scalability: Need 5 more channels for a seasonal campaign? Provision them in your provider's portal and configure your PBX — done in minutes, not weeks.
- Number portability: Keep your existing phone numbers when switching providers or moving offices.
- Business continuity: If your office is unreachable, calls can be rerouted to mobiles, a secondary site, or a cloud-based IVR within seconds.
- Unified infrastructure: Voice and data share the same network, simplifying management and reducing the number of services you need to procure.
Requirements for SIP Trunking
Before deploying SIP trunks, ensure you have the following in place:
- Adequate internet bandwidth: Each concurrent call uses approximately 85–100 Kbps (using the G.711 codec) or 30–40 Kbps (using G.729). A 10-channel SIP trunk with G.711 needs about 1 Mbps of dedicated upstream and downstream bandwidth.
- Quality of Service (QoS): Voice traffic must be prioritised over general data to prevent jitter, latency, and packet loss. Configure DSCP markings on your router and switches to give SIP and RTP packets priority queuing.
- Compatible PBX or IP phone system: Your PBX must support SIP trunking. Most modern IP PBX platforms — 3CX, FreePBX, Yealink P-Series, Cisco CUCM — have native SIP trunk support.
- Session Border Controller (SBC): For larger deployments or where security is paramount, an SBC sits between your PBX and the internet to handle NAT traversal, encryption (TLS/SRTP), and call admission control.
- Firewall rules: SIP (port 5060 UDP/TCP or 5061 for TLS) and RTP (a range of UDP ports, typically 10000–20000) must be permitted through your firewall.
SIP trunks are a common target for toll fraud — attackers who compromise your PBX and make thousands of dollars' worth of international calls. Always use strong SIP authentication credentials, restrict outbound dialling by destination, and monitor call detail records for unusual patterns.
SIP Trunk vs Hosted PBX vs UCaaS
SIP trunking is one of several options for modern business communications. Here is how it compares to hosted PBX and UCaaS (Unified Communications as a Service):
SIP Trunk vs PSTN/ISDN vs Hosted PBX
| Feature | SIP Trunk | PSTN / ISDN | Hosted PBX / UCaaS |
|---|---|---|---|
| Infrastructure | On-premises PBX + internet | On-premises PBX + carrier lines | Cloud-hosted — no on-site PBX |
| Upfront Cost | PBX hardware or software | PBX hardware + line installation | Minimal — subscription-based |
| Monthly Cost | Per-channel or unlimited plans | Per-line rental + call charges | Per-user subscription |
| Scalability | Add channels instantly | Order new lines (days to weeks) | Add users instantly |
| Control & Customisation | Full — you manage the PBX | Full — you manage the PBX | Limited to provider's feature set |
| Maintenance | You maintain PBX hardware/software | You maintain PBX + carrier manages lines | Provider handles everything |
| Best For | Businesses with existing PBX investment | Legacy environments not yet migrated | Small businesses or those wanting zero on-site management |
Pros and Cons of SIP Trunking
Pros
- Significant cost reduction on line rental and call charges
- Instant scalability — add or remove channels without physical changes
- Keep full control of your PBX and call routing
- Business continuity — reroute calls during outages in seconds
- Support for DID numbers across multiple sites on a single trunk
Cons
- Dependent on internet quality — poor connectivity means poor call quality
- Requires QoS configuration and network expertise
- Security risk if PBX or SBC is not properly hardened
- Emergency calling (000/112) may require address registration with the provider
- On-premises PBX still requires hardware maintenance and updates
Frequently Asked Questions
Yes. Number porting allows you to transfer your existing phone numbers from your current carrier to your SIP trunk provider. The process typically takes 5–15 business days in Australia and is managed by the gaining provider.
There is no fixed limit — it depends on your subscription and your available bandwidth. Each concurrent call uses one channel. You can start with a small number and add more as needed without any physical changes.
Not directly. Analogue phones connect to the PBX via FXS ports or an analogue telephone adapter (ATA). The PBX then communicates with the SIP trunk provider over IP. So your analogue handsets can still be used, but they connect to the PBX, not to the SIP trunk itself.
If your internet connection fails, inbound calls cannot reach your PBX. Most SIP providers offer failover options such as redirecting calls to a mobile number, a secondary site, or a voicemail service. For mission-critical telephony, consider a redundant internet connection (e.g., 4G/5G backup).
