01202 612 000

SUPPORT PORTAL

STATUS.VU

Wholesale

STATUS.VU

SUPPORT PORTAL

01202 612 000

SIP Trunks.

Highly-scalable and flexible, easily capable
of managing hundreds of channels.

Take your telephony to the next level.

The 2025 Switch Off will mean your existing ISDN and PSTN legacy ‘landlines’ will soon go silent. But don’t worry. Alongside our own Tier-1 voice network, we have a range of future-proof solutions to keep your team productive from everywhere.

Depending on your existing PBX (Private Branch Exchange), SIP Trunking could be the most cost-effective way to future-proof your telephony, while minimising any disruption to your day-to-day processes.

SIP Trunks.

SIP Trunking

Businesses large and small can benefit from the flexibility of SIP Trunking, enabling cost-effective calling that’s future-proof and easy to scale, as your business grows.

SIP Lite

A bite-sized, two-channel SIP service that’s ideal for small businesses that simply need to replace existing PSTN lines before they’re deactivated at the end of 2025, and nothing more.

SIP Unlimited

Perfect for growing businesses, and those with seasonal surges in demand that need to manage hundreds of channels at any given time, especially call centres and large headquarters.

Why choose SIP Trunking?

R
Enhance the customer experience and customise workflows for better customer satisfaction.
R

Maximise your investment with the perfect upgrade for PSTN-based PBX systems with minimal upheaval.

R

Depending on your existing solution, SIP Trunking could save you money compared to now!

Solid solutions built to succeed.

Future-proof telephony

An internet-based communication solution that enables you to better manage costs long into the future.

Easy management

Customise the caller and staff experience through a simple web portal.

Bird’s eye view

Keep on top of productivity with in-depth reports, no matter where staff are working.

Super scalable

Add new channels in just a couple of clicks, supporting increasing demand, as your business grows.

SIP Trunking technical FAQ.

What is SIP Trunking?

SIP trunking uses the internet to connect a Private Branch Exchange (PBX) system to the public switched telephone network (PSTN) for making and receiving telephone calls.

How are Voip Unlimited’s SIP Trunking solutions unique?

Firstly, we have the best MOS scores around – that means our customers can enjoy the best audio quality, and worry less about whether calls will drop halfway through!

Another thing that makes our SIP solution even more unique is Bundle Plus! As of writing, we believe we’re the only SIP Provider to be able to offer this to the Channel…

We also support a wide range of technologies to deliver and enhance our SIP solutions. Some of the technologies may be new to you, so check out the explanations below for a little walkthrough of how they benefit the solutions you deliver to end-customers.

What technologies are Voip Unlimited’s SIP Trunking solutions compatible with?
We future-proof our communications as much as possible, ensuring support with as many technologies as possible, including:

Technology

Supported?

TCP Support

Yes

UDP Support

Yes

Fax Support

Yes - T38 Bypass

SIP TLS Support

Yes

SRTP Support

Yes

IPv4 Support

Yes

IPv6 Support

Yes

NAPTR / SRV Support

Yes

Auto Failover

Yes

Temporary call diverts

Yes

Inbound outbound call logs

Yes

Advanced Fraud Management

Yes

What is ‘Bundle Plus’?

Our Bundle-plus SIP Trunks are an evolution of SIP (Session Initiation Protocol) Trunking that allows for a certain number of simultaneous voice calls to be made and received simultaneously over a single SIP trunk.

In traditional SIP trunking, a separate channel is required for each voice call that is made or received. So if your customer has 10 channels, they can make any number of inbound or outbound calls (up to a combined total of 10) at any one time. However, this can lead to inefficient use of minute bundles for organisations that need to make and receive a high volume of simultaneous calls.

Let’s say the same customer purchases 10 Bundle Plus channels…

You’ll be able to deliver 10 inbound and 10 outbound channels, for a total of 20 simultaneous calls. This greatly reduces the complexity and cost of setting up and managing multiple SIP trunks, helping you deliver even greater value to your customers.

Why do some SIP solutions, like Voip Unlimited’s, support IPv6 and others don’t?
The main advantage of using IPv6 in SIP trunking is eliminating the need for Network Address Translation (NAT), which reduces the cost of deploying SIP solutions.

NATs map private IPv4 addresses to public IPv4 addresses, allowing organisations to leverage private addresses on internal networks and still be able to communicate with the public internet. But they can cause issues in SIP Trunking.

When SIP messages traverse a NAT, the IP address and port information may be unknowingly modified, which in turn may cause SIP messages to be rejected, dropped or fail to establish a proper connection.

Using IPv6 in SIP trunking eliminates the need for NAT, helping to ensure that SIP traffic is delivered correctly and efficiently. Not only reducing the potential for issues and cost of deployment, but increasing customer satisfaction by providing a more reliable, scalable and efficient SIP solution.

What is NAPTR?
NAPTR (Naming Authority Pointer) records are a type of DNS record that can be used to help route SIP (Session Initiation Protocol) traffic for resilient SIP trunking, providing a more flexible, resilient and robust method of routing SIP traffic for SIP trunking.
What are the advantages of using NAPTR records?
One of the main advantages of using NAPTR records with SIP trunking is that they allow for more flexibility in routing SIP traffic. NAPTR records can be used to specify different priority levels for various SIP servers, as well as specify different types of SIP services (such as voice, video, or instant messaging) that a particular server can handle.

This allows organizations to have more control over how SIP traffic is routed to and from their PBX systems, and can be useful for load balancing and failover management purposes.

Additionally, NAPTR records can also support the use of different transport protocols (such as TCP and UDP) for different types of SIP services. This can be useful for customers that want to use different transport protocols for different types of SIP traffic, such as using TCP for voice traffic and UDP for video traffic.

What are SRV records?
SRV (Service) records are a type of DNS record that can also be used to help route SIP (Session Initiation Protocol) traffic for SIP trunking. Like NAPTR records, SRV records can be used to specify different priority levels for different SIP servers, as well as to specify different types of SIP services (such as voice, video, and instant messaging) that a particular server can handle.
What are the advantages of using SRV records?
One of the main advantages of using SRV records with SIP trunking is that they provide a more direct method of routing SIP traffic. Unlike NAPTR records, which use a series of “regular expressions” to match SIP traffic to a particular server, SRV records use a simple naming convention to specify the location of a particular SIP service, making it easier for organisations and Managed Service Providers to set up and manage SIP Trunks.

Another advantage of using SRV is how they can provide more reliable and efficient routing of SIP traffic, thanks to their simple naming convention. This accelerates the routing of SIP traffic to the most appropriate server and ensures it’s delivered to the correct server pronto, to minimise delays and dropped calls.

They also include backup or failover options, enabling you to specify different priority levels for various SIP servers, so that SIP traffic gets to an available server if a node fails.

What is SRTP?
SRTP is an extension to the RTP (Real-time Transport Protocol) that encrypts the RTP data and provides a mechanism for validating the authenticity of RTP packets, which helps to prevent eavesdropping and tampering with SIP traffic.

The encryption and authentication provided by SRTP can help to protect SIP traffic from various types of threats, such as eavesdropping, man-in-the-middle attacks, and replay attacks.

SRTP also provides several advantages over other encryption protocols, such as RTPsec, ZRTP, and SDES, which are often used in SIP trunking. SRTP is simpler to configure and use, as it doesn’t require the use of separate keys for encryption and authentication, enabling the same key for both encryption and authentication.

What is T.38 Bypass?
T.38 is a standard protocol for sending faxes over IP networks. When a fax call is made over SIP Trunks, the data is first converted from analogue to digital – which can lead to some errors before the data is sent – as a series of T.38 packets.

However, the T.38 protocol isn’t perfect and can cause faxes to fail, or reduce the quality of the fax, as well as being sensitive to network jitter, latency and packet loss. To overcome this issue, T.38 Bypass allows faxes to be sent over a SIP trunk in their original analogue form, bypassing the need for T.38 protocol and minimising the chance of errors.

However, T.38 Bypass is not always appropriate – it depends on the fax machine, PBX and SIP carrier. Also, T.38 Bypass may not be the best solution if there are any QoS issues within a customer’s network, and sticking with T.38 would most likely be a better option.

Your success is our success.

Delivering business-grade connectivity solutions that ‘just work’ isn’t easy, but we make it look simple.
We’re known in the industry for solving issues others won’t even take on, which is why we get feedback like:

Be unique. Be loyal. Be reliable.

Those three mantras guide everything we do at Voip Unlimited.

Unique.

Communication is the lifeblood of any business, so we’ve optimised our network, reducing packet loss and latency while prioritising call traffic to deliver the best voice quality and reliability around.

Loyal.

Thousands of happy customers prove how loyal and dedicated we are. Expect honesty, transparency and loyalty in every interaction, so you can better trust the recommendations we make.

Reliable.

By owning our own private, Tier-1 network, we have more control over performance, keeping one eye on reliability so we meet our business-grade SLAs. And 24/7 support is always there if need-be.

So, what are you waiting for?

Book a call

Want to know more?

If you’re more of a reader than a talker, have a look at one of our recent blogs to discover tips to help enhance productivity, save costs and deliver a better experience for all!

Sales recruitment drive

Voip Unlimited (VU) has expanded its Sales and Account Management team with the addition of two new staff members. The new specialists are using their industry experience focus on delivering VU’s...

read more

Want to work with the best?

Communication is the lifeblood of any business, and when you can’t stay in touch, you lose sales. We make it simple for you and your team, using the most advanced technology to remove challenges to your business growth.

Whether your needs are straightforward or complex, we want to hear from you.

Book a meeting