What Is SIP Trunking? The Plain English Version
SIP stands for Session Initiation Protocol. A SIP trunk is a virtual phone line that connects your phone system to the telephone network using the internet instead of physical copper wires. The word 'trunk' comes from traditional telephony, where a trunk line was the main connection between a business PBX and the phone company's exchange. SIP trunking does the same job, just over your internet connection instead of dedicated copper.Here is the simplest way to think about it. Your business has a phone system (a PBX) that handles call routing, voicemail, hold music, extensions, and ring groups internally. That PBX needs a way to reach the outside world so people can call your business number and your staff can dial out. SIP trunking is that connection. It is the bridge between your internal phone system and every other phone number in the world.Each SIP trunk (also called a SIP channel) represents one simultaneous phone call. If your business needs the ability for 5 people to be on the phone at the same time, you need 5 SIP channels. If 3 staff are on calls and a fourth customer calls in, the call still connects because you have capacity. If all 5 are in use and a sixth call comes in, that caller either gets a busy signal or goes to a queue, depending on how your PBX is configured.How SIP Trunking Actually Works
The technical flow is straightforward. When someone dials your business number, the call hits your SIP provider's servers first. The provider routes it to your PBX over the internet. Your PBX answers the call and sends it to the right extension, ring group, or auto-attendant based on your call flow rules. Outbound calls work in reverse: your PBX sends the call to the SIP provider, which connects it to the destination number over the public phone network (PSTN).The voice data travels as small packets over your internet connection, the same way a video call or a file download does. This is why internet quality matters for SIP trunking. Unlike hosted VOIP where the provider manages everything in the cloud, SIP trunking connects to a PBX that you own or control. That PBX can be a physical box in your office (on-premise) or a virtual PBX running on a server you manage. The key distinction is that you are responsible for the PBX, and the SIP provider is responsible for delivering calls to and from it.SIP Trunking vs Hosted VOIP: The Key Difference
This is the question most businesses actually need answered, and it is where the confusion usually starts. SIP trunking and hosted VOIP both use the internet for phone calls, but they are fundamentally different products aimed at different situations.Hosted VOIP (also called hosted PBX or cloud PBX) means the provider runs everything. The PBX is in their data centre. You plug phones into your network and they connect to the provider's cloud platform. You manage users and settings through a web portal. The provider handles all the infrastructure, updates, security, and call routing backend. You pay per user per month. This is the right choice for most small businesses with 1 to 30 staff. See the hosted PBX vs on-premise PBX comparison for a detailed breakdown.SIP trunking means you run your own PBX and buy the phone line connections separately. You are responsible for configuring the PBX, maintaining it, updating it, and securing it. The SIP provider just delivers the call channels. You pay per trunk (channel) per month, not per user. This gives you more control but requires more technical knowledge.Think of it like renting a fully furnished apartment (hosted VOIP) versus buying a house and connecting the utilities yourself (SIP trunking). The house gives you more control, but you are responsible for the plumbing.When You Need SIP Trunking (And When You Do Not)
SIP trunking makes sense when:- You already have an on-premise PBX and want to replace your ISDN or analog lines with internet-based connections
- You run a self-hosted PBX platform like 3CX, FreePBX, or Asterisk and need carrier connectivity
- Your business has complex call routing requirements that a hosted platform cannot customise deeply enough
- You have IT staff (or an MSP) who can manage and maintain the PBX
- You need a large number of simultaneous call channels (20+) and want to control costs tightly
- You operate multiple sites and want to centralise call management on a single PBXSIP trunking does NOT make sense when:
- You have 1 to 10 staff and no in-house IT capability
- You do not currently have a PBX and do not want to manage one
- You want a plug-and-play solution where the provider handles everything
- Your call routing needs are straightforward (ring groups, voicemail, basic auto-attendant)
- You want to avoid ongoing maintenance, patching, and configuration workFor most small businesses in the 1 to 20 seat range, hosted VOIP is the better fit. It is simpler, lower maintenance, and the monthly cost difference is minimal once you factor in the time and expertise needed to manage a PBX. SIP trunking becomes cost-effective and practical at scale, or when specific technical requirements demand it. If you are unsure which path fits your situation, the phone system sizing wizard can help you work it out.
How SIP Trunk Pricing Works
SIP trunking pricing is different from hosted VOIP pricing. With hosted VOIP, you pay per user per month and everything is included. With SIP trunking, you pay for the trunk channels (how many simultaneous calls you can make) plus call charges on top.Typical SIP Trunk Pricing Components
Trunk rental: This is the monthly fee for each SIP channel. In Australia, expect to pay $5 to $15 per channel per month depending on the provider and volume. Some providers bundle channels into packages (e.g., 5 channels for $40/month, 10 for $70/month). Others sell individual channels. Bulk discounts are common above 10 channels.Call charges: Most SIP providers charge per-minute rates for calls. Local and national calls typically run $0.05 to $0.12 per minute or $0.10 to $0.25 per call (flagfall plus per-minute). Mobile calls are more expensive, usually $0.12 to $0.22 per minute. Some providers offer unlimited local/national calling bundles for an additional flat fee per channel. Always check whether your provider's 'unlimited' plan truly includes mobile calls or just landline-to-landline.DID numbers: DID (Direct Inward Dialling) numbers are the phone numbers people call to reach your business. You need at least one, and most businesses have several (main line, fax, direct lines for key staff). DID numbers typically cost $2 to $8 per number per month. Porting existing numbers in usually has a one-off porting fee of $10 to $50 per number. See the number porting guide for the full process.Setup fees: Some providers charge a one-time setup or activation fee of $50 to $200. Many waive this during promotions or for larger deployments. Always ask.For a practical cost estimate based on your team size and call volume, try the VOIP cost calculator.SIP Trunking vs ISDN: Why Businesses Are Switching
If you have been running ISDN lines (ISDN2, ISDN10, ISDN30), SIP trunking is the direct replacement. ISDN used dedicated copper circuits to deliver phone channels. SIP trunking delivers the same channel capacity over your internet connection. The switch is happening across Australia because the copper PSTN network is being decommissioned. New ISDN connections have not been available for years, and existing services are being progressively disconnected.The practical differences matter. ISDN2 delivered 2 channels. The SIP equivalent is simply 2 SIP trunks. ISDN10 delivered 10 channels, replaced by 10 SIP trunks. The mapping is direct. But SIP trunking adds flexibility that ISDN never had: you can scale channels up or down without waiting for Telstra to provision a new physical circuit, add channels in minutes instead of weeks, and pay significantly less per channel.A typical ISDN2 line in Australia cost $50 to $80 per month for 2 channels. Two SIP trunks from a specialist provider cost $10 to $30 per month for the same capacity. The savings are real and immediate. For larger businesses migrating from ISDN30, the per-channel cost reduction is even more pronounced.How Many SIP Trunks Does Your Business Need?
The number of SIP trunks you need equals the maximum number of simultaneous phone calls your business needs to handle at any one time. This is not the same as the number of staff, the number of phones, or the number of extensions. It is about concurrent call capacity.A general rule of thumb: for every 3 to 4 staff members who regularly use the phone, you need 1 SIP trunk. An office of 10 people where everyone takes calls throughout the day probably needs 3 to 4 trunks. A reception-heavy business where calls come in bursts (medical practice, real estate agency, trades business) may need more trunks relative to staff count because of peak-time call clustering.Here is a rough sizing guide:1-3 staff: 1-2 SIP trunks
4-10 staff: 2-4 SIP trunks
11-20 staff: 4-8 SIP trunks
21-50 staff: 8-15 SIP trunks
50+ staff: 15+ SIP trunks (consider burst capacity for seasonal peaks)Most SIP providers let you add or remove trunks on a monthly basis, so you are not locked in. Start conservative and add capacity if you notice callers getting busy signals or long queue times. Your PBX call logs will show concurrent call peaks, which is the definitive way to right-size.
SIP Trunking Technical Requirements
SIP trunking has a few technical prerequisites that you need to confirm before signing up with a provider.Compatible PBX
You need a PBX that supports SIP trunking. Most modern PBX systems do, including 3CX, FreePBX, Asterisk, Yeastar, Grandstream UCM, and virtually all enterprise-grade platforms (Cisco, Avaya, Mitel). If you have an older analog-only PBX that only connects via ISDN or analog lines, you will either need to replace it or use a SIP gateway (an adapter box that converts between SIP and the legacy PBX interface). Check the PBX explainer for a breakdown of PBX types.Internet Connection Quality
Each SIP trunk (active call) uses approximately 80 to 100 Kbps of bandwidth in each direction when using the G.711 codec (the standard for high-quality voice). The G.729 codec uses less bandwidth (around 30 Kbps per call) but with slightly reduced audio quality. For 5 simultaneous calls on G.711, you need at least 500 Kbps of reliable, low-jitter upload and download bandwidth dedicated to voice.Bandwidth is rarely the bottleneck. What actually kills SIP trunk call quality is jitter (variation in packet arrival time) and packet loss. Jitter above 30ms causes choppy audio. Packet loss above 1% makes calls sound robotic or causes dropouts. A standard NBN 50 or NBN 100 connection has more than enough raw bandwidth, but if that connection is shared with staff streaming, downloading, or backing up to the cloud, the voice traffic will suffer. Use the VOIP bandwidth calculator to check whether your current connection can handle the load.Quality of Service (QoS) Configuration
QoS is a router setting that prioritises voice traffic over other internet traffic. Without QoS, a large file download by one staff member can cause call quality issues for everyone. With QoS enabled, voice packets get priority and the download slows slightly instead. Most business-grade routers support QoS. Consumer-grade ISP-supplied routers often do not, or have limited QoS options. This is one of the practical reasons businesses running SIP trunks often invest in a proper business router.Firewall and NAT Considerations
SIP traffic uses specific ports (typically UDP 5060 for signalling and a range of UDP ports for audio). Your firewall needs to allow this traffic. NAT (Network Address Translation), which almost every business router uses, can cause issues with SIP because of how it handles the two-way audio stream. The most common symptom is one-way audio: you can hear the caller but they cannot hear you, or vice versa.The fix is usually straightforward: disable SIP ALG (Application Layer Gateway) on your router. SIP ALG is a router feature that tries to 'help' SIP traffic pass through NAT, but it frequently corrupts SIP packets instead. Turning it off fixes the majority of one-way audio and registration issues. Your SIP provider's support team will typically walk you through this during setup.SIP Trunking Security
SIP trunking exposes your phone system to the internet, which introduces security considerations that ISDN and analog lines never had. The most common attack is toll fraud, where hackers gain access to your PBX and make thousands of international calls at your expense. This is not theoretical. It happens to poorly secured PBX systems regularly, and the bills can be tens of thousands of dollars.Protection is straightforward if you follow basic steps:- Use strong, unique passwords for all SIP accounts (not admin/admin or 1234)
- Restrict SIP registration to your provider's IP addresses in your firewall
- Enable fail2ban or equivalent brute-force protection on your PBX
- Set international call barring on your trunks unless you specifically need international dialling
- Monitor call logs for unusual patterns (calls at 3am, calls to premium-rate numbers)
- Keep your PBX firmware and software updatedMost reputable SIP providers also offer fraud protection features: daily spend limits, international call alerts, and automatic trunk lockout if unusual activity is detected. Ask about these when evaluating providers.
Australian Businesses: SIP Trunking Specifics
SIP trunking works the same way globally, but there are several Australia-specific factors that affect your deployment.NBN Connection Types and SIP Performance
Your NBN connection type directly affects SIP trunk reliability. Fibre to the Premises (FTTP) and Fibre to the Building (FTTB) deliver the most consistent performance for voice. Fibre to the Node (FTTN) can work well but is more susceptible to line quality issues because of the copper segment between the node and your premises. Fixed Wireless NBN adds latency and jitter that can affect call quality during peak congestion. Satellite NBN is generally unsuitable for SIP trunking due to latency (600ms+ round trip makes conversation impractical).Upload speed matters more than download for SIP trunking because both sides of a call need real-time data flowing. NBN 50 plans deliver 20 Mbps upload, which is plenty for even 20+ simultaneous calls on paper, but real-world upload performance can dip during peak hours on FTTN connections. If you are running more than 5 SIP trunks on NBN, consider a business-grade internet plan with a service level agreement (SLA) that guarantees minimum speeds.Number Porting to SIP Trunks
When you switch to SIP trunking, you will almost certainly want to keep your existing phone numbers. Number porting in Australia is regulated by the ACMA and typically takes 5 to 10 business days for standard geographic numbers. Porting 1300 and 1800 numbers can take longer, sometimes 10 to 15 business days, and requires the Responsible Carrier to approve the transfer. During the porting process, your numbers continue to work on the old service until the cutover moment, so there is no gap in reachability.The critical step is to NOT cancel your old service before the port is complete. Cancelling early can release the number back to the carrier pool, making it much harder (sometimes impossible) to recover. Let the port complete, confirm calls are landing on your new SIP trunks, and then cancel the old service.1300 and 1800 Numbers on SIP Trunks
If your business uses a 1300 or 1800 number, these work seamlessly on SIP trunks. Your SIP provider can port the 1300/1800 number to their platform, and incoming calls route to your PBX just like any other call. The one difference is that 1300/1800 numbers have a separate 'Responsible Carrier' designation, so porting involves an additional step compared to standard numbers.Emergency 000 Calling
SIP trunks support 000 emergency calling, but with an important caveat. Unlike traditional landlines, VOIP-based 000 calls may not automatically transmit your physical address to emergency services. Your SIP provider should allow you to register a physical address against your account for 000 location purposes. Confirm this is set up correctly during deployment, especially if your business has staff who may need to call 000 from the office.Power Outages
SIP trunking relies on your internet connection, your router, and your PBX all having power. If the power goes out, your phones go down. Traditional PSTN lines were powered from the exchange and worked during blackouts. This is a real consideration for businesses in areas with unreliable power. Solutions include an uninterruptible power supply (UPS) for your router and PBX (provides 30 to 90 minutes of backup), configuring your PBX to failover calls to mobile numbers during outages, or using a hosted PBX component that can reroute calls to mobiles when the on-premise system goes offline.AU SIP Trunk Provider Landscape
Australia has a healthy market of SIP trunk providers serving businesses of all sizes. The major categories are:Specialist VOIP providers: Companies like Maxotel, VoIPLine, and Symbio that focus on business VOIP and SIP services. These tend to offer better pricing per trunk, more flexible configurations, local AU support, and deeper technical assistance for PBX integration. They are the best fit for small to medium businesses.
Major carriers: Telstra, Optus, and TPG all offer SIP trunking products. These suit larger enterprises that want everything under one provider umbrella and are willing to pay premium pricing for the brand. Expect higher per-trunk costs, longer contract terms, and less hands-on technical support for PBX configuration.
Wholesale/carrier-grade: Providers like Symbio and Vocus sell SIP trunking at wholesale level, often through MSPs and IT resellers. If you work with an MSP that manages your IT infrastructure, they may provision SIP trunks through one of these carriers on your behalf.
What Most Businesses Get Wrong
1. Choosing SIP trunking when hosted VOIP would be simpler and cheaper. The most common mistake is buying SIP trunks because they sound more 'professional' or 'serious' without having the PBX infrastructure or technical knowledge to support them. If you do not already have a PBX and someone who can manage it, hosted VOIP gives you the same call quality and features with none of the maintenance burden. SIP trunking is not inherently better than hosted VOIP. It is a different product for a different situation.2. Underestimating internet quality requirements. Businesses buy SIP trunks, plug them into a shared NBN connection with no QoS, and then wonder why call quality is terrible during busy periods. SIP trunking works brilliantly on a properly configured network. It works terribly on a congested one. Budget for a business-grade router with QoS if you are running more than 2 to 3 trunks on a shared connection.3. Not disabling SIP ALG on the router. This single setting causes more SIP trunking issues than any other factor. One-way audio, dropped calls, registration failures, calls cutting out after 30 seconds. Disable SIP ALG on your router before you even start configuring SIP trunks. It should be step one of every SIP deployment.Your Next Steps
If you are considering SIP trunking, work through this checklist before contacting a provider:1. Confirm you actually need SIP trunking. Do you already have a PBX, or are you planning to deploy one? Do you have technical staff or an MSP to manage it? If the answer to both is no, hosted VOIP is likely the better path.2. Determine how many trunks you need. Count the maximum number of simultaneous calls your business handles at peak times. Add 1 to 2 for headroom. That is your starting trunk count.
3. Check your internet connection. Run the bandwidth calculator to confirm your connection can handle the load. Check your NBN connection type and upload speed.
4. Confirm your PBX supports SIP trunking. Check the manufacturer documentation or ask your IT support. Most modern PBX platforms do.
5. Get pricing from 2 to 3 providers. Compare per-trunk cost, call rates, DID number fees, and contract terms. Ask about fraud protection features and porting support.
6. Plan your number porting. Identify all numbers that need to port. Do not cancel old services until porting is complete and confirmed.If you are not sure whether SIP trunking or hosted VOIP is the right fit, get a free recommendation and we will help you work it out.
What is the difference between a SIP trunk and a SIP channel?
In practice, the terms are used interchangeably. Technically, a SIP trunk is the overall connection between your PBX and the SIP provider, and a channel is one simultaneous call path within that trunk. When providers sell 'SIP trunks,' they are usually selling individual channels. If you buy 5 SIP trunks, you can make 5 simultaneous calls.
Can I use SIP trunking without a PBX?
No. SIP trunking is specifically a connection between a PBX and the telephone network. Without a PBX, there is nothing for the SIP trunk to connect to. If you do not have a PBX and do not want to manage one, hosted VOIP (where the provider runs the PBX for you in the cloud) is the right option.
Is SIP trunking cheaper than hosted VOIP?
It depends on your scale. For small businesses (1 to 10 users), hosted VOIP is usually comparable or cheaper when you factor in the cost of owning, maintaining, and securing your own PBX. For larger businesses (20+ users) with existing PBX infrastructure and IT staff, SIP trunking typically costs less per user because you are only paying for trunk channels, not per-user licenses.
What happens to my SIP trunks if the internet goes down?
Your phone system goes down with it. SIP trunks rely entirely on your internet connection. You can mitigate this by configuring failover routing on your PBX (redirect calls to mobile numbers when the system is unreachable), using a UPS to keep your equipment running during short power cuts, or having a backup internet connection (e.g., 4G/5G failover).
Can I port my existing phone numbers to a SIP trunk provider?
Yes. Number porting to SIP trunk providers works the same as porting to any other carrier in Australia. Standard geographic numbers take 5 to 10 business days. 1300 and 1800 numbers may take 10 to 15 business days. Your provider handles the paperwork. The critical rule is to not cancel your existing service until the port is confirmed complete.
Do I need a static IP address for SIP trunking?
It depends on your provider and PBX configuration. Some SIP providers require a static IP to whitelist your PBX for security. Others support dynamic IP registration. A static IP is generally recommended for business SIP deployments because it simplifies firewall rules, improves security, and avoids re-registration issues when your IP changes. Most business-grade internet plans include a static IP.
What codecs should I use for SIP trunking?
G.711 (also called PCMU or PCMA) delivers the best voice quality and is the standard for business SIP trunking. It uses about 80 to 100 Kbps per call. G.729 uses less bandwidth (about 30 Kbps per call) with slightly reduced quality, suitable if bandwidth is limited. Most providers support both. Start with G.711 and only switch to G.729 if bandwidth is a genuine constraint.
Can I mix SIP trunking with hosted VOIP?
Some businesses run a hybrid setup where the main office uses SIP trunks connected to an on-premise PBX, and remote workers or branch offices use hosted VOIP handsets or softphones from the same provider. This is a valid approach, but it adds complexity. Discuss this with your provider before committing to ensure the two systems integrate cleanly.
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