In a nutshell: A SIP trunk is the modern replacement for old copper / ISDN phone lines — instead of a physical pair of wires from the telco, you get a logical "trunk" delivered over your existing data network. It connects your PBX (or cloud calling platform) to the public phone network using the SIP protocol. SIP trunks are cheaper per channel, easier to scale, and a prerequisite for any modern unified communications deployment. Migration is mostly about numbers (porting DIDs without dropping calls) and quality (sizing bandwidth, configuring QoS, putting an SBC in front).
What you're replacing: PSTN, ISDN, analogue
Before SIP, businesses bought voice connectivity in three main shapes:
- Analogue lines (POTS). One copper pair = one voice channel. Cheap, simple, used for everything from a hotel guest room to a small office's main line. Limited features.
- ISDN BRI (Basic Rate Interface). 2 voice channels per circuit. Common for small offices in the 1990s–2000s.
- ISDN PRI (Primary Rate Interface). 30 voice channels per E1 circuit (Europe/Asia) or 23 per T1 (US). The workhorse of mid-market and enterprise PBXs for two decades.
All three are still around but in decline. IMDA and the Singapore carriers are progressively retiring legacy PSTN infrastructure; major carriers have communicated their plans to end-of-life ISDN and POTS over the coming years. Most "new" PSTN connections sold today are already SIP underneath, with an analogue terminal adapter (ATA) or gateway converting them at one end.
What a SIP trunk is
A "trunk" in telephony just means a connection between two switches that can carry multiple simultaneous calls. A SIP trunk is the same idea, but the switches talk to each other using SIP (Session Initiation Protocol — see VoIP Fundamentals) and the calls are carried as RTP audio streams over an IP network.
Two important properties:
- "Channels" become licences, not wires. You're billed for the number of concurrent calls you're entitled to carry, not for physical lines. Need to handle 50 simultaneous calls instead of 30? Buy more channels — no truck roll, no rewiring.
- Numbers are decoupled from circuits. Your DIDs (Direct Inward Dial numbers — the +65 6xxx xxxx range you publish to customers) are now logical assignments held by your carrier, not wired to specific lines. Move offices, change providers, redirect numbers in software.
Anatomy of a SIP trunk
From the customer side, a typical SIP trunk deployment looks like:
- Your PBX (on-prem like 3CX, Cisco UCM, Avaya, Mitel, FreePBX — or cloud like Microsoft Teams Phone, RingCentral, 8x8) generates and accepts SIP signalling.
- A Session Border Controller (SBC) sits between the PBX and the carrier. The SBC handles security, NAT traversal, protocol normalisation, and call admission control. See SBCs Explained for the detail — it's a layer you almost certainly need.
- The carrier's SIP trunk endpoint — usually identified by an IP address or hostname you configure on the SBC.
- Your internet or private link carries SIP and RTP between the SBC and the carrier. Many carriers offer their SIP trunks over a private network (separate from your public internet) for predictable quality.
For cloud calling platforms (Teams Phone, Zoom Phone, Webex Calling) the SBC may live in the cloud and be managed by the vendor or partner. The pattern is the same.
Why migrate
- Cost. Per-channel cost is typically lower than ISDN, and international and long-distance call rates are usually much lower because the carrier can route them over IP.
- Elastic capacity. Scaling up (or down) is a billing change, not a physical change.
- Geographic flexibility. SIP trunks can deliver numbers from many cities to one PBX, useful for businesses that want local presence in multiple markets without separate PBXs.
- Modern features. SIP trunks support HD voice (G.722, Opus), video, fax-over-IP (T.38), and presence — features ISDN can't carry.
- Cloud-readiness. Any move to a cloud calling platform (Teams Phone, Zoom Phone) ultimately needs SIP trunks or a fully cloud-based PSTN service.
- The decommissioning clock. Carriers will eventually stop offering ISDN. Plan a migration before yours is forced.
Who you can buy from in Singapore
- Major carriers: Singtel, StarHub, M1, MyRepublic — all offer SIP trunking with Singapore DIDs, often bundled with managed PBX or UC services.
- Independent specialists: ViewQwest, Bluetel, Crystone, Voiceflex (regional) — pure-play voice and SIP trunk providers with competitive pricing and more flexible commercial terms.
- Global UCaaS providers: Microsoft (Operator Connect / Direct Routing), Zoom Phone, RingCentral, 8x8 — combine the SIP trunk and the calling platform into a single per-user subscription.
- International CPaaS: Twilio, Vonage, Plivo — programmable voice / SIP trunks, useful for contact centres, IVR, or apps where voice is part of the product rather than just employee telephony.
For directory listings of Singapore telecom providers and integrators, see the telecommunication category on TechDirectory.
How SIP trunks are priced
Most providers price along three dimensions:
| Dimension | What it means | Typical pricing |
|---|---|---|
| Trunk / channels | How many simultaneous calls you can carry | Per-channel monthly fee, or bundled into a per-user UCaaS price |
| DID numbers | Phone numbers presented to the outside world | Per-number monthly fee; first-batch may be bundled |
| Outbound minutes | What you pay for calls made | Local: per-minute or bundled. Mobile and international: per-minute, varies by destination |
Different providers slice this differently — some bundle everything into a flat per-user price, some unbundle aggressively. When comparing, normalise to "cost per concurrent call channel per month" and "outbound minute cost for your typical destination mix."
Migration playbook
- Audit current state. List every line you have (PSTN, ISDN BRI, ISDN PRI, mobile-tied numbers), every DID number, every PBX feature people use, every fax machine, every alarm panel and lift line.
- Decide your destination. Cloud calling (Teams Phone / Zoom Phone) vs keep-the-PBX-but-replace-ISDN. The destination shapes everything else.
- Pick a carrier and SBC. Get test trunks and sample DIDs. If using a cloud calling platform, check whether the carrier is on the "Operator Connect" or equivalent certified list.
- Bandwidth and QoS. Each concurrent G.711 call needs ~100 kbps each way; HD codecs more. Size headroom into your WAN, and configure QoS to prioritise voice packets. See LAN vs WAN Basics for why this matters.
- Pilot with new numbers first. Run the new SIP trunk in parallel with the old ISDN for a few weeks using fresh DIDs (or test extensions). Validate call quality, codec negotiation, voicemail, transfers, conferencing, fax, and any IVR flows.
- Number porting. This is the riskiest step. Plan ports in batches, schedule outside business hours, confirm the carrier's rollback plan, and brief stakeholders that occasional calls may not connect during the cutover window.
- Decommission legacy. Don't leave old ISDN circuits live longer than needed — you're paying for them. But keep written confirmation of the port-complete date before instructing cancellation, or numbers can vanish.
- Document. The new system is software-defined and easy to change — and easy to break. Document trunk configs, dial plans, SBC rules, and emergency-call routing so someone other than the installer can troubleshoot.
Pitfalls & mistakes
- Forgetting the orphans. Fax machines, lift emergency phones, alarm panels, EFTPOS terminals — they often rely on analogue PSTN. Inventory them before cutover and provide ATAs or alternatives.
- Skipping the SBC. Most reputable carriers require one. Even if they don't, security and interoperability headaches that follow are worse than the SBC's cost.
- Under-engineered emergency calls. 995 / 999 must route correctly with the registered service address. Test before go-live and retest after any topology change.
- No fallback path. A single internet line going down takes your phones with it. Pair with a second WAN link (different carrier) or a mobile failover, or use a carrier-provided private link instead of public internet.
- "Lift-and-shift" without rethinking. Migration is the right moment to clean up dial plans, retire unused DIDs, and rethink call flows — not just to replicate ISDN-era patterns in IP.
Where to go next
- The gateway layer: Session Border Controllers Explained — step 3 of the Voice Modernisation path.
- Refresher: VoIP Fundamentals if you want the SIP/RTP basics in more depth.
- Pick a vendor: our Telecom Provider buyer's guide.
Related articles:
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