SIP Trunking
Connect your IP PBX to the public telephone network.
Keep the phone system you already run and move its calls onto a SIP connection. Your PBX stays in charge of extensions and routing; the trunk carries calls to and from ordinary phone numbers.
At a glance
- Connects
- Your IP PBX to the public telephone network
- Calls
- Inbound and outbound
- Capacity
- Sized by concurrent calls
- Signalling
- SIP, with TLS where supported
Compatibility is confirmed for your PBX before anything is agreed.
What it is
A phone line, delivered over IP.
An IP PBX is the phone system that runs your extensions, menus and internal calls. On its own, it cannot reach the outside world: it needs a connection to the public switched telephone network (PSTN), the network every ordinary phone number belongs to.
A SIP trunk is that connection. Instead of physical phone lines, your PBX exchanges calls with Muqatel over an IP connection using SIP (Session Initiation Protocol) to set calls up and tear them down, with the voice carried as a separate media stream. Muqatel connects those calls to the telephone network.
The problem it solves
Traditional lines tie capacity to physical circuits at a specific location, and changing them takes time. With SIP trunking, capacity is set in channels, numbers are not bound to one building, and the connection uses the internet access you already manage. That can reduce reliance on traditional lines; the actual effect on cost depends on your current setup and calling patterns.
Core capabilities
What a SIP trunk gives your PBX.
The exact configuration is agreed with you and confirmed in writing.
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Inbound calls
Calls to your numbers are delivered over the trunk to your PBX, which routes them to extensions, menus or queues as it does today.
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Outbound calls
Your PBX places calls to ordinary phone numbers through the trunk. Destinations and rates are agreed in your quotation.
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Channel capacity
The trunk is sized by concurrent calls: how many conversations can run at the same moment, inbound and outbound together.
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Number assignment
Existing numbers can be ported where supported, or new numbers assigned, and mapped to the trunk.
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Caller ID presentation
Outbound calls can present your business numbers, within the rules that apply to each destination country.
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Security options
Authentication by registration or by trusted IP address, and encrypted signalling and media where both sides support it.
Use cases
When SIP trunking is the right fit.
SIP trunking suits businesses that want to keep control of their own phone system.
- 01
Replacing traditional lines on an existing PBX
Your IP PBX stays in place; the analogue or digital lines feeding it are replaced by a SIP connection.
- 02
Bringing several sites onto one connection
Offices running their own PBX can use SIP trunks rather than separate local lines at each location.
- 03
Connecting a contact centre platform
Contact centre software that supports SIP can take inbound and place outbound calls through a trunk sized for its peak.
- 04
Adding capacity as you grow
Because capacity is set in channels, it can be reviewed when your call volumes change, subject to your agreement.
How it works
From first conversation to live calls.
Timelines depend on your PBX, your network and any number porting, so they are agreed with you rather than fixed in advance.
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Step 1: Review your PBX and network
We look at your PBX platform and version, the codecs it supports, your internet connection and your firewall setup.
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Step 2: Agree channels, numbers and settings
We size concurrent calls from your busiest periods, plan numbers and porting, and agree caller ID and security settings.
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Step 3: Configure the trunk
We provide connection details and work with whoever manages your PBX to set up registration or IP authentication.
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Step 4: Test, then go live
Inbound and outbound calls are tested, including call quality and number presentation, before traffic moves over.
Compatibility and technical considerations
What to check before you connect.
These are the points that decide whether a SIP trunk works well. We go through each of them with you or your PBX administrator.
- Concurrent call capacity
- Count the calls happening at the same time at your busiest moment, not the number of staff. Too few channels means busy signals for callers; too many means paying for capacity you do not use.
- Codecs
- G.711 (A-law or ยต-law) is widely supported and gives good quality. Other codecs may be available depending on the platform. Both your PBX and the trunk must share at least one codec; mismatches can cause one-way or failed audio.
- Bandwidth
- A G.711 call uses roughly 90 kbps in each direction once packet overhead is included. Multiply by your concurrent calls and leave headroom for other traffic on the same connection.
- Firewall and NAT
- SIP signalling and the RTP media streams must pass through your firewall in both directions. Router features such as SIP ALG often rewrite SIP messages and can break calls; they usually need to be disabled.
- Encryption
- TLS can protect signalling and SRTP can protect the audio, where both your PBX and the service support them. We confirm what is available for your setup.
- Caller ID policies
- You may only present numbers you are entitled to use, and some countries restrict or validate caller ID on incoming calls. Presentation can therefore differ by destination.
- Emergency calling
- Emergency calls over IP depend on your internet connection and power, and the caller location may need to be registered. Arrangements and limitations differ by country and are explained before go-live.
- Compatibility
- Most current IP PBX platforms support SIP trunking, but not every device, version or configuration is compatible. Compatibility is confirmed for your specific system before anything is agreed.
Works with
How SIP Trunking fits with our other services.
SIP trunking connects a phone system you run. The other two services cover the phone system itself and the numbers people dial.
What is SIP trunking?
SIP trunking connects a business phone system (an IP PBX) to the public telephone network over an IP connection instead of traditional phone lines. SIP (Session Initiation Protocol) sets up, manages and ends calls; the voice itself travels as a separate media stream, usually RTP.
With a SIP trunk in place, your PBX can place outbound calls to ordinary phone numbers and receive inbound calls on the numbers assigned to it.
Can I connect my existing IP PBX?
In many cases, yes. Most current IP PBX platforms support SIP trunks, but compatibility depends on the specific system, its software version, the codecs it supports and how your network is set up.
Tell us which PBX you use and how it is deployed. We will confirm whether it can be connected and what configuration it needs before anything is agreed.
What is the difference between Cloud PBX and SIP trunking?
SIP trunking supplies the connection between a phone system you already run and the telephone network. You keep managing the PBX itself.
Cloud PBX is the phone system, hosted for you. Extensions, call routing, IVR menus, queues and voicemail are configured online, and staff use desk phones, softphones or apps without an on-site PBX to maintain.
If you have a PBX you want to keep, SIP trunking is usually the starting point. If you would rather not run one, Cloud PBX is usually the better fit.
What do I need technically to get started?
For any of the services you need a stable internet connection with enough bandwidth for your expected number of simultaneous calls, and a network that does not interfere with SIP or voice traffic. Some routers apply a feature called SIP ALG that can disrupt calls and may need to be disabled.
For SIP trunking you also need a compatible IP PBX. For Cloud PBX you need supported devices: IP desk phones, softphones on computers, or mobile apps, depending on the package. We will go through the specifics for your setup.
Can I make emergency calls?
Emergency calling over IP-based services has limitations that differ by country. It can be affected by power or internet outages, and the location passed to emergency services may need to be registered in advance and kept up to date, particularly for remote or mobile users.
We will explain the emergency calling arrangements and limitations that apply to your service and location before it goes live. Please keep an alternative way to reach emergency services where required.
What are concurrent calls or channels?
Concurrent calls (sometimes called channels) are the number of calls a SIP trunk can carry at the same moment, inbound and outbound combined. It is separate from the number of users: twenty staff may need far fewer than twenty channels if they are rarely all on the phone at once.
Looking at your busiest periods helps size this correctly. We can help estimate it from your current call patterns.
Find out whether your PBX is ready for SIP.
Tell us which PBX you run, how many calls you handle at your busiest, and where you call. We'll confirm compatibility and suggest a configuration.