OnSIP

First, let’s address SIP: Session Initiation Protocol is the de facto standard protocol for establishing, conducting and ending a VoIP call. If you are really interested in technical details, check the Wikipedia entry on SIP.

SIP is to VoIP as SMTP is to E-mail. Just like the standard protocol, SMTP, which allows two email servers to exchange email data, SIP allows two endpoints (IP Phones) to connect to one another using a standard protocol. Without such a standard, two phones would not share a common set of instructions and guidelines in order to properly exchange packetized voice.

The great thing about SIP is that it promises to help do away with phone numbers all together. How? Imagine your email address could be “dialed” from a phone and it would kick off a call. Further, imagine everyone at your company is reachable by phone using email addresses. My name is Robert. It would be great if people could reach me by dialing Robert@xxxxxx rather than some random collection of digits, which have no meaning.

Think of SIP as the protocol for yet another service capability of a domain:

Email: SMTP
Web: HTTP
Voice: SIP
Video: SIP

OnSIP is a SIP service provider. With an OnSIP account, you can open your domain to SIP traffic. This will allow you and your team members to be reached by phone using email address too!

You can make and receive SIP calls to/from any SIP address. Calls to users on our network, since they are all SIP calls, are all free.

Check out our SIP Domain Features for more info.

Many people ask me what a virtual phone number is. Well, everyone knows what a phone number is. So the real question is “What is Virtual about a Virtual Phone Number?”

First, here is a dramatically oversimplified history lesson:

Historically, phone numbers were tied to physical locations. The phone company would provision a phone number to work over a single physical line, which would be “dropped” at the actual location the number would be tied to. Calls to that number could only be delivered to that physical location and businesses would have to receive the calls using expensive PBX systems which maintained routing smarts, voicemail applications, IVRs, etc.

With a virtual phone number, the physical limitation is removed, allowing a company to use a phone number in a more flexible manner with no reliance on physical presence of phone lines or phone systems. Calls to a virtual phone number are handled by a remote agent or proxy, which forwards on calls based on user defined rules. This allows a business to:

• Seamlessly connect multiple locations
• Eliminate on-premises telco equipment, telco space, phone lines, etc.
• Maintain phone service during incidents effecting physical offices.

Here is an example of how a business uses a virtual phone number from OnSIP:

Company X maintains a New York headquarters and a Los Angeles sales office. The company has local phone numbers and one toll free number, all virtual phone numbers. When a customer calls the toll free number, rather than having it answered by a phone system in either the Los Angeles or New York office, an IVR answers the call on the OnSIP Virtual PBX service. When prompted by the IVR, the caller selects option 2, for sales. Because there are sales associates in both offices, phones ring simultaneously in both offices until answered in Los Angeles. The call is from a key customer who needs to speak to the CEO who is working from his beach house in Cape Cod. The sales associate transfers the call to the CEO who is connected to OnSIP using his home office cable Internet connection. When the call is completed, the CEO uses 4-digit dialing to a make a free call to the sales associate in Los Angeles to congratulate her on a job well done for helping close a major sale.

The entire team is connected via OnSIP, which acts on behalf of the users, no matter where they are now or where they move. Users have the flexibility to make and receive calls and use the service as if they were in the office at all times.

OnSIP has phone numbers available throughout the country and are available for immediate activation.

Did you know that Junction Networks has a public web service API available to any of our customers that exposes all of the pieces necessary to manage your own hosted PBX and PSTN gateway services? In fact our admin.onsip.com web administration portal has been built entirely on top of the very same API that is open and available to the public - this means any feature you see in our administration portal is potentially available for you to implement in your very own VoIP product.

What does this mean for your business?

This means that you could build your own web portal to extend your product line to include a VoIP service without having to do the work of building the VoIP side of the service! You would be able to leverage all of the features from our OnSIP Hosted PBX and PSTN gateway products into your own product behind your own custom web portal. For example, let's say that you are a web host looking to augment your product offering in order to differentiate yourself from your competitors. By implementing the various pieces of the Junction Networks web service API you would allow your own customers to create and manage their own hosted PBX. You could even do some more advanced things like host your customers SIP domain in the primary domain that he has registered with your service already. Simply add an SRV record to identify Junction Networks as the domain's SIP service provider and the customer will be able to get their web, email, and voice service all from you - while you don't need to worry about any of the voice end of things. Additionally, you can implement your own pricing structure on top to fit your new VoIP offering to be in line with your current pricing. Since OnSIP never charges for users or extensions you can offer your implementation either with or without such charges - the choice is yours.

The benefits of implementing your new VoIP product offering this way is that you can continue to concentrate on your core business without a need to dedicate permanent valuable resources to building and maintaining your VoIP product. That's why we're here! It is our job to make sure that your VoIP services continue to stay up and working reliably.

Getting Started with the web service API

To get started with the Junction Networks API you can start by trying it out yourself on our API demo page. In order to use the demo you need to be a registered Junction Networks customer with an authentication name and password, you can signup for a free 30 day trial here to obtain these credentials. You can also get started with our VoIP web service API by reading the API documentation. If you happen to be looking for a feature that is not currently documented then please send us a support request and we'll be sure to get the API call documented as soon as possible.

Have fun.

One of the reasons why I love VOIP so much is its ability to unify technology and simplify my life. I've gotten used to the conveniences of getting voice mail delivered to my e-mail and being able to turn my laptop into my desk phone and work from my pajamas, but I still want more.

One of the newer features that has really spoiled me is our Firefox Click-to-Call Add-on, which allows me to dial any number listed on a web page from my configured SIP phone with a single mouse-click. Why waste time typing in a phone number? VOIP can save you the effort and you can spend your time doing what a human is really necessary for -- actually having the conversation.

Where does the Firefox Add-on become particularly useful? Any web page with numbers on it now becomes a dialing directory, which means that your Intranet with the phone directory of all of your employees is now a time-saving tool. If you use a web-based CRM, like Salesforce, calling a customer is as simple as clicking on the phone number in their account. (We like this feature a lot.) Want to order a pizza? Call a restaurant and make reservations? You're likely to go to the restaurant's web page to find the number, so why not let your computer also dial the number for you?

Cell phones made it unnecessary to punch in a number years ago, so why still do it on your desk phone?

Hopefully you like the changes to our marketing WWW Sites, http://www.junctionnetworks.com and http://www.onsip.com. Our marketing department is right on track with our messaging. We are using the Drupal content management system which should allow us more timely updates to the site, our knowledgebase and our Blog.

In addition to the new marketing WWW sites, we have also launched a new User Portal. The User Portal allows all OnSIP Hosted PBX users to log into the system to see their settings. Previously, only account administrators were able to log into the interface. The User portal only contains information pertaining to that user and does not show any account-level information. The new User Portal also contains a newly revamped and updated knowledgebase.

While we were at it, we launched a new application: Conference Bridge for OnSIP. OnSIP users pay $19.95 per month for unlimited conference bridge use for up to 10 (ten) simultaneous callers. Any callers coming in via the PSTN would still pay the regular 2.9c/min.

As an update, we recently changed the pricing on our ACD Queues to $19.95 for the ability to have up to 5 callers in a queue simultaneously. Around the same time we launched “Announcements” under the “Apps” tab. Announcements allow you to play a recording to the caller and then send the caller to a new destination. This is useful for ‘business hours’ or ‘directions’ announcements within an IVR application. Lastly, we launched the “Inbound Bridge” application which allows a customer to purchase phone numbers (DID) for any other VoIP provider and ‘bridge’ those DIDs into the Junction Networks service.

The next big release for us will be the User Dashboard, but more about that later.

Please let us know what you think about our new WWW Sites and our new applications.

SIP stands for session initiation protocol and its adoption as an open-source communications standard is revolutionizing communications. Within the Junction Networks universe SIP addresses control everything. Each username, extension, voice mail box or auto-attendant is an alias for a unique SIP address.

I know you are wondering “this is very interesting but why should I care? As long as when I pick up my phone and get a dial-tone, and my voice mail works, I am good to go.” This is true, but a SIP address can also be used as a powerful communications tool. My email address is tim@junctionnetworks.com. It is also my SIP address. If you have a SIP phone you can dial that address and call me...for free. You can be anywhere in the world. I can be anywhere in the world. With the next generation of WiFi enabled smart phones supporting SIP addresses I can be sitting in a Starbucks coffee shop in London and you can call me from a Starbucks in Sydney. For free.

Now entering SIP addresses into phones is a little complicated (but no harder than entering in an email address) but when you sync up your phone with Outlook or Thunderbird you can transfer all your contacts information easily enough. Junction Networks offers phone number click-to-call through the Firefox web browser, an ongoing open-source project called Cockatoo (http://cockatoo.mozdev.org/ui.html) is doing the same thing for the Thunderbird email program.

Most Junction Networks users have a SIP address in the following format tim@acme.onsip.com. However with a few simple changes to your domain names SRV records you can easily have your SIP address match your email address. Junction Networks is working with several leading cell phone manufactures to test various smart phones for true SIP capabilities. We will be adding reviews and updating our progress on the blog. These are really exciting times for the phone industry and SIP is playing a leading role.