SIP Trunks.

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.

We future-proof our communications as much as possible, ensuring support with as many technologies as possible, including:

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.

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.

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.

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.

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.

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.

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.

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.