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June 2002

Tom Keating

The SIP Coolness Factor

BY Tom Keating

Not long ago, there were proprietary little VoIP gateway boxes targeted at consumers that allowed you to make point-to-point calls between them across the Internet. These boxes were great little gadgets, especially for individuals with a loved one or friend that lived far away, which often resulted in high monthly phone charges. By placing one of these units on each end, such as one from Aplio (www.aplio.com) or Innomedia (www.innomedia.com), one could save on long-distance charges. Calling from unit-to-unit was often free and only when you hopped off to the PSTN did you actually have to pay a discounted per minute rate. The problem with these boxes was that they were proprietary; for example, you couldnt call from an Aplio unit to an Innomedia unit. Worse, they didnt have a real phone number associated with them just an IP address, which meant you couldnt call the unit from a PSTN line. Essentially, these SOHO VoIP gateways were a one-way street; you could call out, but no one could call in after all, what phone number would they dial?

Several of these SOHO VoIP gateways added H.323 support in their future releases, but they still didnt have a phone number mapped to the units. Then came Session Initiation Protocol (SIP) the perfect protocol to enable service providers to provide not just voice services over IP, but also enhanced services, including unified communications and even the ability to assign a real phone number to SIP-based products.

One interesting company leveraging the power of SIP to provide unified communications across the Internet is Webley (www.webley.com). Webleys Media Switching Platform (MSP) is an open architecture software-based platform that enables carriers to quickly offer new services, such as voice mail, speech-recognition virtual assistant, and more. In addition to SIP, it supports many interface protocols, such as RTP, ISDN, SNMP, POP3, WAP, IMAP4, SMTP, XML, LDAP, and VPIM.

WorldCom and Webley
Webley claims that their MSP platform can be scaled to support millions of users. The fact that WorldCom has chosen Webley to offer SIP-based voice services certainly supports this claim. The carrier is deploying SIP voice mail servers from Webley on its vBNS+ network, where all IP Communications, call signaling, and handoffs to the PSTN take place. Worldcoms product, called IP Communications, negates the need for a $50,000+ CPE-based PBX. All that is required are SIP-based endpoints connected to the LAN at the customer premises, such as Ciscos 7960, Pingtels xpressa, or Siemens omniPoint 100. Since SIP phones are often a bit pricey, in theory you can use Mediatrixs four-port SIP-to-analog gateway and use regular analog phones. In addition you can probably use a SIP-based soft-phone client such as Microsofts MSN Messenger.

Using the SIP protocol, Centrex features are enabled across IP, hence the term IP-Centrex. Adding additional lines, voicemail features, turning on call forwarding, etc. can all be done simply via a Web browser. Barry Zipp, senior director of Enhanced Voice Services at WorldCom was quoted as saying, To support native SIP you have to replicate and enhance the functions that are resident in a PBX or Centrex environment, which is what WorldCom is doing, he says. Business users can expect features such as call forwarding, call transfer and four-digit dialing.

In any event, I had Webley send me a beta account to test their platform. I was provided with the account information to access the Web interface and the settings to put in the SIP-based MSN Messenger client. They assigned me a toll-free number, which when dialed has the capability of call blasting me at five possible endpoints. These endpoints can be any traditional phone, such as a cell phone, office phone, etc.; but the real beauty of Webley is that it supports SIP endpoints. Thus, all I had to do is enter my SIP address sip:tkeating@messenger.webley.net:5080 as one of my possible endpoints and Webley would attempt to reach me on any SIP client I am currently using. The endpoints are simply entered via a Web browser to a password-protected Web site.

The Features
Webley includes call transfer rules that are based on caller ID and current date and time. Similarly, a specific voice mail greeting can be played to callers based on caller ID and current date and time. Webley also features a natural speech-recognition-driven virtual assistant. Other features include a unified messaging mailbox for voice mail, faxes, and e-mail, voice-activated dialing, follow-me service via one number, conference calling, and the capability to allow users to listen to e-mail over the phone. Via a personal Web page, customers can upload and synchronize PIM address books, retrieve/forward messages and set up conference calls.

Since MSN Messenger is free and I dont need to lug any hardware (i.e., a SIP hard phone) around, I can log on to Webleys SIP registrar using MSN Messenger and receive incoming calls to my toll-free number. Very cool! In fact, by using MSN Messenger in conjunction with Webley, I am able to receive and make calls wherever I am on my office PC, my home PC, in a hotel using a laptop, etc.

Make The Call
When I tested it, I made a test call to my toll-free number, and it asked me if I wanted to find him or take a message. After selecting find him it prompted me to say my name, which it recorded. Next, my cell phone and office phone simultaneously rang. In addition, on my PC I received a MSN Messenger instant message popup saying 2032650243@ would live to have a video and voice conversation with you. Do you want to accept or decline? If I answered the call on a PSTN line, it plays back the recorded screened name and I can then accept the call simply by saying Ill take the call or state Send to voice mail.

Additionally, depending on where I answered and accepted the call (cell phone, office phone, or MSN Messenger), I am instantly connected to the caller while the other legs of the call are disconnected. Obviously, the coolest way of accepting the call was via MSN Messenger, which I tested, and it performed flawlessly.

Quality and Latency
The voice quality was some of the best Ive heard on a VoIP connection and the latency was exactly the same as a cell phone. I know because I made a call from my cell phone to my toll-free number, which routed the call to MSN Messenger and I answered it with my PCs USB phone. I then held my cell phone to my left ear and a USB phone to my right ear. I spoke some test words into both receivers simultaneously and I heard it played back at the exact same moment on both receivers. What this means is that the Webley VoIP service has just as good latency as a cell phone network, which is pretty impressive (see sidebar).

SIP & Telecommuting
According to the International Telework Association and Council, last year, mobile workers numbered 28 million in the U.S., 17 percent more than a year earlier. With the economic slowdown businesses are turning to telecommuting because of the cost savings. By utilizing SIP, businesses can offer their employees an inexpensive means of working from home while still retaining the full functionality of the corporate PBX.

While I see Internet telephony products all the time, I was still awed by Webleys coolness factor. I firmly believe that before long we will all want the ability to receive phone calls wherever we are. SIP is the enabling technology to that end. It wont be long before the carriers support routing all your phone numbers (home, office, cell phone, etc.) to a SIP-based client that you are currently logged onto. Using presence technology that is inherent in SIP, the carriers will know exactly where to route your calls at anytime or according to scheduling rules that you define. Perhaps these advanced features and services will be just what the doctor ordered to help the telecom sector break out of its slump.

Tom Keating is CTO of Technology Marketing Corporation, and the executive technology editor of TMC Labs. He can be reached at Tom Keating .

[ Return To The June 2002 Table Of Contents ]

Credit Where Its Deserved

I have to give Microsoft credit for vastly improving the latency of their SIP-based VoIP client, which is much better than NetMeeting. Although I must admit Microsoft still has a way to go. MSN Messenger still does not support traversing NAT firewalls. They say they are working on it, but Ive heard through the grapevine that their answer is Stop using Linksys or D-link firewalls and just use Microsofts Personal Firewall and it will work fine. That is, however, questionable. Supposedly Universal Plug and Play support on NAT firewalls will also solve this problem, but who knows whether or not existing firewalls out there will be upgradeable to this new standard. Thus, I still believe that Microsoft should make MSN Messenger more NAT-friendly.

Its certainly not impossible for SIP to traverse NAT firewalls. Vonage (www.vonage.com) uses Ciscos SIP-based ATA 186 units and their proprietary SIP-thru-NAT technology makes Vonages DigitalVoice accessible on any and all broadband connections. Coincidentally, TMC Labs reviewed Vonage in this issue. Also, check out Packet IN  for more on this subject.

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