Save money with SIP Trunking
SIP Trunking from Aria VoIP offer a reliable, cost-effective and feature rich replacement for ISDN Circuits and analgue lines. While ISDN circuits have traditionally been favoured as a way to make and receive calls to/from your PBX, availability of high speed Data Connections, as well as increased stability in IP Platforms have led to SIP Trunks becoming a preferred option for call routing to/from the BT PSTN network.
Compared to ISDN circuits, SIP Trunks can provide far great functionality at a lower cost. SIP Trunks are instantly scaleable and provisioned in real-time. When bundled with an advanced fraud prevention service, it’s unsurprising that a vast number of businesses are migrating from traditional fixed line services to a digital one.
In order to make and receive calls through your SIP Trunk, your IP-PBX will connect to your on-site Network (generally through the Router). The IP-PBX will then communicate with our Call Routing platform over a Data Connection (such as Broadband, EFM or a Leased Line) in the form of millions of synchronous data packets. Calls can be served to your SIP Trunk in HD (high-definition) and are not limited to just voice, video and instant messaging is growing in popularity.
When you make an outbound call through your IP PBX, that call travels through your Data Connection to reach our Call Routing platform, which then ‘hands-off’ the call to the upstream networks. When receiving a call from the PSTN network, the call reaches our Call Routing platform, which then forwards the call to your IP-PBX through the SIP Trunk(s).
Your inbound and outbound calls will follow bespoke call routing patterns, according to rules configured in our easy-to-use web portal. The web portal allows for easy initial configuration, as well as real-time changes to your office phone system if required. Because the Call Routing platform is hosted off-site, and accessible online at any time, you do not need to book an engineer if you wish to make any changes. Because our Technical support team is available 24/7, and conveniently based in our Norwich offices, help is available if you have any difficulty with these changes.
SIP Trunks are virtual connections between your IP-PBX and our VoIP network. SIP or the Session Initiation Protocol, is the standard communications protocol for voice and video in a Unified Communications (UC) solution using an internet or broadband connection. A SIP trunk replaces the need for traditional analog service that utilises the PSTN or POTS (plain old telephone system).
A SIP trunk allows connections to scale up or down in real-time based on channels. Channels is simply another way of describing concurrent calls or the number of lines for voice or video. The only major consideration in understanding a channel limit is the bandwidth available on the internet connection at any one time.
Aria ViIP Powered by VoiceHost SIP Trunks are RFC3261 compliant.
VoiceHost is a Tier 2 Telecommunications provider, which means we operate our own Call Routing infrastructure and telephone number ranges. However, we do not maintain the National Infrastructure (commonly known as the PSTN network) used to deliver the calls themselves. Instead, we use the existing infrastructure (through multiple interconnects) managed and maintained by Tier 1 Telecommunications providers. All calls (besides on-net calls to other users on the VoiceHost system) utilise the extensive British Telecom PSTN network, ensuring optimal call quality, reliability and scope.
At VoiceHost, we consider ourselves a different sort of Telecommunications company. Our background is in software development, not Traditional Communications. This means we approach problems and solutions in a very different way to most Telecommunications companies. We have built, and continue to develop on, an agile, scalable and highly intuitive platform for Advanced Communications.
This means a few things for our customers:
Our platform is intuitive and easy to configure. It’s been designed to be used by real customers without a background in Telecoms, not just engineers.
Our SIP Trunking solutions are deployed from the same control panel as Hosted PBX features, giving you all the tools you need to build a highly advanced hybrid solution for your business. SIP Trunks and Hosted PBX functionality can be fused together, opening up possibilities that previously never existed.
The platform functionality continues to evolve as the needs of our customers evolve. We’re constantly adding features that have been requested by our customers, and looking for new ways to make the lives of our customers easier.
We use the same systems for provisioning as our customers. This means faster help when you need it, without the technical jargon.
Because we’re not reselling another platform, every change you make to your configuration is instant. We’ve done away with the days of waiting weeks for a minor config. change.
For most businesses implementing a VoiceHost SIP Trunking solution, there may be some capital investment, and some installation required. Every customer who requires SIP Trunking, requires the following:
- A reliable Internal Network
- A reliable Data Connection
- A properly configured IP-PBX capable of using our SIP Trunks
- A properly configured SIP Trunk on the VoiceHost Portal