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.
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.
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.
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?
Maximise your investment with the perfect upgrade for PSTN-based PBX systems with minimal upheaval.
Depending on your existing solution, SIP Trunking could save you money compared to now!
Solid solutions built to succeed.
An internet-based communication solution that enables you to better manage costs long into the future.
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.
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?
Yes - T38 Bypass
SIP TLS Support
NAPTR / SRV Support
Temporary call diverts
Inbound outbound call logs
Advanced Fraud Management
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?
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?
What are the advantages of using NAPTR records?
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?
What are the advantages of using SRV records?
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?
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?
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.
Be unique. Be loyal. Be reliable.
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.
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.
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?
Want to know more?
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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.
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