What is SIP trunking?
SIP trunking is a service that connects a company's phone system to the public telephone network over the internet. Instead of buying physical phone lines, you rent a number of simultaneous calls, called channels, from a provider. Your phone system, known as a PBX, a contact-centre platform or an application, sends calls to the provider across an IP connection, and the provider connects them to ordinary landlines and mobiles anywhere.
The name comes from two ideas. SIP, the Session Initiation Protocol, is the signalling language that sets calls up and tears them down. A trunk is a shared link carrying many calls at once, the way a trunk road carries many cars. Put together, a SIP trunk is a shared internet link that carries many voice calls between your system and a carrier.
Companies use it to replace old ISDN lines, to cut calling costs, to add capacity during a busy season and to give a contact centre local numbers in many places. Our SIP trunking and voice products work this way, and you can add virtual numbers to the same trunk.
How SIP trunking works
A phone call needs two things: a way to agree that a call should happen, and a way to carry the sound.
- Signalling (SIP). Your PBX sends an INVITE message to the provider saying who is calling whom. The provider answers with progress messages such as ringing, then connected, and later a BYE message to end the call. The rules are in RFC 3261.
- Media (RTP). Once the call is connected, the audio travels as small packets using the Real-time Transport Protocol, usually over a different range of ports from signalling.
- Codecs. Both sides agree how the sound is compressed. G.711 is uncompressed quality at 64 kbps, G.729 uses about 8 kbps and suits limited bandwidth, and wideband codecs such as Opus sound better where supported.
When someone calls out, your PBX sends the invitation to the provider's SIP address. The provider checks who you are, checks that the destination is allowed and that you have credit, then hands the call to the right carrier. Incoming calls work in the other direction: a number you own on the provider's network rings your PBX.
SIP trunking vs PRI and traditional lines
| SIP trunk | PRI (ISDN) | Analogue line | |
|---|---|---|---|
| Delivery | Over your internet connection | Dedicated digital circuit | Copper pair |
| Capacity | Add or remove channels quickly | Fixed blocks, 23 or 30 channels | One call per line |
| Cost | Lower per channel; pay for what you use | Higher; monthly circuit fee | High at scale |
| Numbers | Any location, many numbers, easy to move | Tied to the local exchange | Tied to the line |
| Resilience | Reroute if a site fails | Needs a second circuit | Needs spare lines |
| Best for | Most new voice deployments | Sites with weak internet | A single fax or alarm |
The PRI is not obsolete; it still fits places where the internet is unreliable or equipment cannot speak SIP. For everything else, the ability to scale channels in minutes is the advantage.
What you pay for
Pricing varies between providers, but most charges fall into the same four groups.
| Charge | How it works | How to keep it down |
|---|---|---|
| Channels | A monthly fee per concurrent call you reserve | Size to the busiest hour, not to headcount |
| Minutes | Per-minute rates by destination: local, national, mobile, international | Check rates for your top five destinations |
| Numbers (DIDs) | A monthly fee per inbound number | Release numbers you no longer advertise |
| Extras | Call recording, emergency calling, number porting fees | Ask which are included |
Compare providers on the all-in cost of your actual call mix, not on a single headline rate. The pricing page lists ours, and the cost calculator lets you model volumes. Ask too whether calls are billed by the second or in rounded blocks.
Mind the caller ID. Calls that show a wrong or unregistered caller ID can be blocked or flagged as spam. Present a number you own, in international format, and read our guide on SIP trunk setup, codecs and caller ID before you go live.
Benefits and trade-offs
- Lower cost. You pay for channels and minutes used rather than fixed circuits, and calls between your own sites cost nothing extra.
- Elastic capacity. Handle a sales campaign or an exam-results week by adding channels for a month, then taking them back.
- Number flexibility. Local numbers in a city where you have no office; numbers that move with you if you relocate.
- Resilience. If an office loses power, route calls to another site or to mobiles within minutes.
And the trade-offs, so you can plan for them. Voice quality depends on your internet connection, so a trunk over a congested link will crackle. You also become responsible for firewall rules and security, because a SIP port open to the world will be probed within hours. Use IP allow-listing, strong credentials or TLS, and limit international dialling by default.
How to set up a SIP trunk, step by step
- Count your concurrent calls. Look at your busiest hour in the last three months and add some headroom. That is the number of channels you need.
- Check your network. Make sure your upload and download can carry 100 kbps per call with spare capacity, and that jitter and packet loss are low. A quick test call from the office at peak time says more than a speed test.
- Choose how you authenticate. Use IP authentication for a PBX with a fixed public address, or username and password registration where the address changes.
- Order and assign numbers. Provision the numbers you need and set the caller ID you will present. If you are moving existing numbers, ask about porting time first.
- Configure your PBX. Add the provider's SIP address, set codecs in your preferred order, enable a jitter buffer and open the firewall for SIP and the RTP port range.
- Test, then migrate gradually. Place test calls in both directions, then send around ten per cent of live calls through the trunk for a day before you move the rest.
For an Asterisk-based PBX, a minimal outbound trunk looks like the sketch below. The hostnames and addresses are placeholders; use the values from your provider's dashboard, and restrict the allowed address range.
[telnxo-trunk]
type=endpoint
transport=transport-udp
context=from-trunk
disallow=all
allow=ulaw
allow=alaw
allow=g729
aors=telnxo-trunk
rtp_symmetric=yes
force_rport=yes
[telnxo-trunk]
type=aor
contact=sip:sip.example-provider.com:5060
[telnxo-trunk]
type=identify
endpoint=telnxo-trunk
match=203.0.113.10If you use another provider now, keep it configured as a fallback for a week. Our migration notes for Exotel and Vonage show how trunks, numbers and callbacks map across.
Common problems and quick fixes
| Symptom | Likely cause | Fix |
|---|---|---|
| One-way or no audio | NAT or a blocked RTP port range | Open the RTP range, enable NAT traversal, turn off SIP ALG on the router |
| Calls drop after 30 seconds | ACK or session timer not reaching the PBX | Check the public IP configured in the PBX and firewall rules |
| Choppy audio | Packet loss or jitter, wrong codec | Prioritise voice traffic, try G.711 on a fast link |
| Calls rejected with 403 | Authentication or IP not allowed | Verify the allowed IP or credentials in the dashboard |
| Calls rejected with 488 | No codec in common | Add G.711 to the PBX codec list |
Use number lookup to check that a number you are dialling is valid and what type it is, and read errors in the error code list. Our analytics show answer rates, call duration and failure reasons per route.
Is SIP trunking right for you?
If your PBX or contact-centre software already speaks SIP and you have decent internet, almost certainly yes. A trunk is also the foundation for adding voice to applications. The same provider can run a click-to-call button on your website, and support teams can combine voice with SMS and WhatsApp. Read how to put them together in our guide to the SMS gateway and the overview of the WhatsApp Business API.
A SIP trunk is only as good as the network between you and the carrier. Test it at your busiest hour, not at midnight.
Need a SIP trunk with local numbers? Tell us your peak concurrent calls and destinations and we will size it with you.
Frequently asked questions
What is SIP trunking in simple terms?
It is a way to connect your business phone system to the regular phone network through the internet instead of physical phone lines. Your system sends call set-up messages and audio to a provider, who connects the call to the person you are dialling.
What is the difference between SIP and SIP trunking?
SIP, the Session Initiation Protocol, is the signalling language used to start, change and end calls. A SIP trunk is a service that uses SIP to link your system to the public telephone network.
How many SIP channels do I need?
One channel carries one call at a time. Count the most calls you have at once during your busiest hour and add around 10 to 20 per cent. Buying channels for every employee is a common and costly overestimate.
SIP trunking vs PRI: which is better?
A PRI is a fixed digital line with a set number of channels, typically 23 or 30. SIP trunking is delivered over IP, so you can add or remove channels quickly and usually pay less. PRI still makes sense where internet quality is poor or equipment cannot support SIP.
How much bandwidth does a SIP call use?
About 80 to 100 kbps each way for a G.711 call including overhead, and around 30 kbps for G.729. Multiply by your peak concurrent calls and keep extra capacity for other traffic.
Is SIP trunking legal in India?
Rules for connecting internet telephony to the public network are specific in India, and depend on how the service is set up and who holds the licence. Check the current requirements with your provider and the telecom regulator before you route live traffic.
Why do I get one-way audio?
Usually a firewall or NAT problem: the audio (RTP) packets are blocked or sent to a private address. Open the RTP port range, enable NAT traversal on the PBX and check that the SIP application layer gateway on the router is off.