In our last post we explained the basics of VoIP, how SIP became the default standard in the VoIP industry, and the fact that even though most VoIP providers have adopted the SIP protocol, they use a proprietary implementation to keep calls within their own network. In this post, we will expand our discussion on SIP, explain SIP URI, why Virtual PBX chose to adopt the open SIP URI standard for VoIP communication, and finally why this gives you the freedom to choose the VoIP solution that works best for you and your business.
In 2009, Virtual PBX broke the mold by expanding our feature set to allow calls to go through the Virtual PBX virtual phone service, but rather than dial out to find the extension owner at a phone number, they could instead have the Virtual PBX service send the call via SIP to their VoIP service of choice that accepts inbound calls via SIP URI.
In our last blog we described SIP (Session Initiation Protocol) as the default protocol on which the VoIP industry has standardized. A SIP URI (Uniform Resource Identifier) is basically an internet “address” for a VoIP service allowing one VoIP user to call another VoIP user by using the SIP URI. A SIP URI resembles an e-mail address and follows the format of email@example.com or user@Ipaddress (e.g. User@192.168.1.7). The “user” portion of the URI could be a user name or resemble a 10- or 11-digit phone number. The “domain” portion is the domain of the VoIP provider and can be a name or an IP address. For instance, CallCentric uses the following format: firstname.lastname@example.org.
There are many VoIP services available which fit this mold and are very cost-effective, including CallCentric, Ekiga, Truphone, and Gizmo5, although after being purchased by Google in 2009, Gizmo5 does not accept any new sign ups. With most of these VoIP services, you download a softphone to your computer, which is a software application that has a dialpad and is used to make and receive phone calls, typically using a USB headset or USB phone. However, some of the above services go a step further and allow the user to purchase a hardware VoIP phone and register it to the service, thus allowing the user to have a phone on their desk. This configuration could allow the user to replace their land line if so desired. With most of these services, users can call each other for free. But users can also get a phone number, or DID, and attach it to their VoIP service allowing users outside their VoIP network to call them as well. This feature will usually add a monthly or per minute cost to the service.
Now that we’ve explained SIP URI and some of the VoIP services that have implemented it, let’s circle back around to why Virtual PBX chose to support it. Through our VoIP Peering feature, for which we have won the Internet Telephony Excellence Award 2 years in a row, users can choose to have the Virtual PBX service reach them on not only their cell phones and land lines, but also send calls via SIP URI directly to the VoIP service of their choice. You can find information on how to do this in our training video.
So why is this so cool? Let’s put the parts together. Not only can a VoIP phone service save you a lot of money over a traditional land line with long distance, but if it’s truly an open SIP solution, then it can work with other VoIP services on the market. All this allows users to choose best of breed VoIP services that truly meet their telecom needs and the needs of their business rather than being tied down by proprietary VoIP solutions. With Open VoIP Peering, Virtual PBX is now the lowest-priced and most flexible hosted PBX solution on the market, allowing users to mix and match their Virtual PBX virtual phone system with the SIP-based VoIP softphone (and/or hardware phone) service of their choice, or with one they were already using. This creates amazing opportunities for businesses to best meet all of their feature, cost and reliability needs. And that’s what happens when vendors like Virtual PBX put the customer first.