Service · SIP Trunking
SIP trunking for the phone system you already run
Connect your IP PBX to the public telephone network over IP. Calls in both directions, numbers routed where you need them, and a configuration that is reviewed, documented and tested before it carries live traffic.
- IP or registration authentication
- Inbound and outbound
- Failover routing
What it is
A voice connection delivered over IP
A SIP trunk is a virtual phone line. Your PBX uses the Session Initiation Protocol (SIP) to set up calls with our platform, and the audio travels as RTP packets over your internet or private connection.
With traditional lines, every call needs a physical circuit to the local exchange. With a SIP trunk, your PBX sends calls to the provider over a data connection you already have, and the provider connects them to the public telephone network. Callers notice no difference; your team keeps the same handsets, extensions and menus.
A trunk is defined by two things that are often confused: the numbers (DIDs) that callers dial to reach you, and the channels that set how many calls can be in progress at once. You can have many numbers on a trunk with only a few channels, or a few numbers with many channels.
Channels and numbers are separate parts of a SIP trunk
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Channels
- What it is
- Capacity for one simultaneous call
- Sized by
- Peak concurrent calls, in and out
- Example
- 10 channels = 10 calls at once
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Numbers (DIDs)
- What it is
- A phone number that routes to your trunk
- Sized by
- How many numbers people need to dial
- Example
- 50 numbers, e.g. one per employee
Why organisations move to SIP
Problems a SIP trunk solves
Most moves to SIP start with one of these limitations of traditional lines.
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Fixed-capacity lines
Analogue lines and ISDN circuits come in fixed blocks. Adding capacity means new physical circuits and lead times; a SIP trunk is resized in configuration.
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Numbers tied to a location
Traditional numbers are bound to the exchange that serves the building. With SIP, numbers can be routed wherever your PBX runs, including a data centre or the cloud.
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Limited resilience
When a physical line fails, calls fail with it. SIP trunks can send inbound calls to alternative destinations when your PBX cannot be reached.
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Separate providers per country
Organisations with offices in several countries often manage separate line contracts. SIP connectivity can bring numbers and outbound routing under one configuration, subject to local availability.
Inbound and outbound
Both directions on one trunk
The same trunk normally carries inbound and outbound calls, configured to your routing rules.
Inbound calls
When someone dials one of your numbers, the carrier network routes it to your trunk and we deliver the call to your PBX in an agreed number format. Your PBX then applies its own rules: a menu, a queue, a ring group, a direct extension or out-of-hours handling. Each number can be routed differently.
Outbound calls
Your PBX sends outbound calls to the trunk, which passes them to the carrier network for delivery. Destinations are enabled to match your needs, and international or premium-rate calling is opened deliberately rather than by default.
Caller ID presentation
Where supported, your PBX can present any number you are authorised to use, such as a main number for the whole company or a direct line per user. Presenting numbers you do not hold is not permitted, and rules vary by country.
Authentication
IP-based or registration-based authentication
Every trunk needs a way to know that calls really come from your PBX. There are two standard methods, and the right choice depends on your network.
Choosing an authentication method
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IP authentication
- How it works
- Calls are accepted only from your PBX’s static public IP address
- Best for
- On-premises PBXs and SBCs with a static IP
- Security
- Strong: traffic from any other address is rejected
- What we need
- Your public IP address(es)
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Registration (digest)
- How it works
- Your PBX registers with a username and password and re-registers periodically
- Best for
- Dynamic IPs, hosted PBXs, or connections you do not control
- Security
- Depends on credential strength and protection
- What we need
- Nothing; we issue the credentials
Network and media
Network, codec and DTMF requirements
These settings decide call quality. Each one is listed on your configuration sheet.
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Codecs
G.711 is the common baseline and avoids transcoding. Compressed or wideband codecs depend on the route and your PBX. We agree a codec order so both sides negotiate the same codec.
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DTMF
Keypad tones must survive the call path so menus, voicemail and payment systems work. RTP events (RFC 4733) are the usual choice; the method must match on both sides.
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Bandwidth
Each concurrent call consumes bandwidth in both directions including packet overhead. Size your connection for peak calls plus your other traffic, and prioritise voice.
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NAT and firewalls
One-way or missing audio is almost always a NAT issue. Disable SIP ALG on routers that mishandle SIP, set the PBX’s public address correctly and open the agreed RTP port range.
Compatibility
Works with standards-based IP PBXs
Asterisk, FreePBX, 3CX, FreeSWITCH, FusionPBX, Yeastar, Grandstream UCM and other SIP-compliant systems can usually connect. We review your platform and version before you commit.
Product names are trademarks of their respective owners and are listed for compatibility guidance only; no partnership or certification is implied.
; Example only. Replace placeholder values with those on your
; configuration sheet. Shown for Asterisk PJSIP with IP authentication.
[provider-trunk]
type = endpoint
transport = transport-udp
context = from-provider
disallow = all
allow = ulaw,alaw
dtmf_mode = rfc4733
aors = provider-trunk
direct_media = no
[provider-trunk]
type = aor
contact = sip:SIP-HOST-FROM-CONFIG-SHEET:5060
[provider-trunk]
type = identify
endpoint = provider-trunk
match = PROVIDER-SIGNALLING-IP/32Reliability and security
Failover, fraud controls and troubleshooting
Resilience and security are agreed during setup, not added after an incident.
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Inbound failover
If your PBX does not respond, calls to your numbers can be sent to an alternate destination: a second PBX, another site, or a mobile number. Agreed in advance and tested.
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Outbound resilience
Outbound calls depend on your PBX reaching the trunk. Resilience comes from a second internet connection, a second site, or a secondary trunk configured on the PBX.
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Fraud prevention
IP allow-lists, strong credentials, channel limits and restricted international destinations reduce the risk and impact of toll fraud. We agree sensible limits during setup.
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Monitoring and troubleshooting
SIP response codes and call traces show why a call failed. We use them during testing and when investigating issues, and share findings with your team.
Is it right for you?
When SIP trunking fits, and when it does not
SIP trunking is usually a good fit if
- You already run an IP PBX (or plan to) and want to keep control of it
- You are replacing analogue, ISDN or PRI lines
- Your call volumes or capacity needs change through the year
- You need numbers in more than one city or country routed to one system
- You want inbound calls to fail over when your site is unavailable
Consider an alternative if
- You do not want to run or manage a PBX: a hosted phone system may suit you better
- Your internet connection cannot carry voice reliably and cannot be upgraded
- You need features only available from your current carrier’s exchange service
Pricing
What determines the price
SIP trunk pricing is quoted against your requirements. Every line of the quote is itemised.
- Number of concurrent channels, and whether you need burst capacity
- Numbers (DIDs) required, by country and type
- Outbound destinations and expected monthly minutes
- Inbound minutes on toll-free or specific number types
- Redundancy: secondary trunks, failover destinations or sites
- Any one-off configuration, porting or integration work
FAQ
SIP trunking questions
How many channels do I need?
Count the most calls that happen at the same moment during your busiest hour, inbound and outbound together, then add headroom for growth. Call reports from your current PBX or carrier bill give the best data. Our guide on estimating channel requirements walks through the method.
Should I use IP authentication or registration?
If your PBX has a static public IP address, IP authentication is simpler and more secure, because only traffic from that address is accepted. If your address changes, or your PBX is behind a connection you do not control, registration with strong credentials is the practical choice.
Can I move my existing numbers to a SIP trunk?
Often yes, through number porting. Portability depends on the country, number type and current provider. We check before you cancel anything. See Virtual Numbers for how porting and new numbers work.
Will a SIP trunk work with my firewall?
Usually, once it is configured for SIP. The key settings are allowing SIP from the provider’s addresses, opening the RTP media port range and disabling SIP ALG where it interferes. We review your firewall model during the compatibility check.
Is emergency calling supported?
Emergency calling rules differ by country and often require a registered address for each number or site. Requirements and any limitations are confirmed for your locations before go-live, and we explain them in writing.
Next step
Get a SIP trunk configured for your PBX
Tell us your PBX platform, peak concurrent calls and the countries you need. We will reply with compatibility notes and an itemised quote.
Helpful to include
- PBX platform and version
- Static IP or dynamic connection
- Peak concurrent calls
- Numbers to port or order