SIP Phone Service Overview
What is SIP Trunking?
SIP trunking is a method of delivering telephone and other unified communications services over the Internet to customers that have SIP enabled private branch exchange (IP-PBX) solutions. SIP utilizes both Voice over Internet Protocol (VoIP) and Session Initiation Protocol (SIP) and it replaces traditional telephone lines or PRIs (Primary Rate Interface).
How do SIP Trunks Work?
Traditional business phone systems consist of two key components. The PBX, which provides call management and features such as Auto Attendants and voicemail, and the PRI lines which connect calls to the PSTN (Public Switched Telephone Network) where they are routed to the destination telephone. When SIP trunks are utilized, the IP enabled PBX connects to the data network instead of the PRI lines and the voice traffic travels the Internet to connect to the PSTN. SIP Trunks can also be used with analog adapters or SIP-to-T1 gateways, allowing you to keep your legacy PBX equipment and take advantage of lower telecom costs.
Why do Businesses Choose SIP Phone Service?
While there are many advantages to the VoIP SIP trunk approach, the primary drivers are cost and flexibility. SIP trunking eliminates the need for PRI lines and the associated cost. Unlike PRI lines, which contain 23 channels, SIP trunks can be purchased in increments as low as one channel, or one concurrent call. This gives businesses the ability to purchase and pay for only what they need and to easily scale as capacity requirements change.
Virtuo Platform Features
Tier-1 Redundant Network
Virtuo uses only Tier-1 upstream providers to rout traffic for our customers. This means that our clients get the best quality voice along with a redundant platform that ensures performance and reliability. We employ multiple gateways across the United States to eliminate any single point of failure.
Powerful Control Panel
We offer a powerful easy-to-use control panel to manage all aspects of your account. You can easily purchase and manage phone numbers; add, change or discontinue service; review and export real-time call data records and modify your billing preferences.
Bring Your Own Bandwidth
Virtuo gives you the flexibility to choose your own broadband Internet connection. Unlike some SIP trunking providers, we do not require that you get data connectivity through us. Choose from cable, DSL, T-1 or Metro Ethernet, whichever works best for you. Each G.711 IP phone call will consume approximately 85kbps up/down across your network. We also allow the G.729 compressed codec which cuts this number down to around 35kbps, but only recommend it in areas where bandwidth is at a premium.
Simple SIP Trunk Pricing
Paying for your SIP VoIP trunk is as easy as setting it up. Our service is prepaid with no-contract and costs we offer a no contract rate of $24.95 per month per unlimited SIP trunk channel. Add channels or cancel anytime. You can easily upgrade channels online in the Control Panel. DIDs are inexpensive and you can search for and purchase DIDs right from the Control Panel.
Widespread DID Coverage
If your business required Direct Inward Dialing (DID), Virtuo offers a large database of telephone numbers across the US in many rate centers. We also offer International DIDs in more than 40 countries around the world. It’s easy to search our database and order numbers and most DID number orders are provisioned instantly. We can even back-order numbers if they are unavailable in your area.
Real-Time Call Data Records
Call Data Records (CDRs) can be useful in understanding your true communications needs and also give you a view of business activities. All of your CDRs are available in the Control Panel and are shown in near real-time. You can easily view today’s calls or search up to a year in the past. CDRs can also be downloaded as a .CSV file.
Dialer Traffic Welcome
Most SIP trunking providers avoid outbound dialer and telemarketing traffic. We have special arrangements with upstream carriers who welcome type of traffic and can accommodate this need. Simplify identify this in your trunk settings to avoid surcharges for dialer traffic on our regular service.
International Toll Fraud Protection
Our systems watch in real-time for International call fraud and take immediate action to kill unauthorized calls in progress, shut off International calling on your trunk and alert both our customer and our Network Operations Center of suspicious activity. These automated systems protect you even when your premise equipment is compromised. While it is not possible to stop 100% of fraud, our client’s have the best protection in the industry.
Fault Tolerant DID Routing
A unique advantage to our systems is the ability to have an inbound DID route to a primary and a secondary IP-PBX. This means if you are running dual PBX’s for redundancy we can first attempt to deliver the inbound call to your primary PBX and if there is a failure, send it to a secondary PBX. We also offer the ability to forward the call to a backup PSTN number in the event of a routing failure. All of this can be seamlessly configured in the Control Panel.
Works Great with Asterisk and Other Open Source Solutions
Our SIP Trunking service is a perfect fit for open source systems such as, Asterisk, FreeSwitch, Elastix, PBX in a Flash and other popular Graphical User Interfaces to configure and control Asterisk. We provide detailed ‘cut-and-paste’ trunk configuration settings enabling you to be up and running on our service in a matter of minutes.
No Obligation Free Trial
We’re so confident that you’ll love our service, we offer free trial accounts which include 60 minutes of outbound calling to anywhere in the US48 to test our service. It’s easy to get started. Once your email is confirmed, we automatically create a SIP trunk for you and deposit test credit. You can then configure it to your system and start making calls immediately.