When the race began for a VoIP signaling/call control protocol in the mid-1990s, one of them got off to a head start. It was H.323, a bloated binary protocol (really a suite of protocols) specializing in conferencing and the more sophisticated telecom applications. It was inspired by the telecom industry, was based on ISDN's Q.931 recommendation, and it had the blessing of the International Telecommunications Union (ITU).
In true tortoise-and-hare fashion, however, 1999 brought the appearance of a new application level, peer-to-peer signaling protocol that would soon dominate the VoIP industry: the Session Initiation Protocol (SIP), the creation of a working group within the Internet Engineering Task Force (IETF), which had labored on it since 1996.
SIP turned out to be everything that H.323 was not: a lean, flexible, text-based protocol more in tune with the design philosophy of other Internet protocols such as the HyperText Transfer Protocol (HTTP). SIP isn't just about call control; it can set up, modify and tear down unicast or multicast multimedia "sessions" that can include voice, video and any other form of media. SIP-enabled end platforms include at least a protocol client (the "user agent client") and server ("user agent server"). A SIP session includes other SIP components such as proxy, redirect, registration and location servers, which resolve names, locate users and relay SIP requests and responses. In small systems all of these can exist as software running in the same box.
With SIP, both the server and client have complete control over their session. In the old circuit-switched PSTN world, the "endpoints" (phones) have no call control capabilities and all services must be delivered and controlled by a big, expensive, central office switch or a corporate PBX. Fortunately, SIP-enabled endpoints can be easily implement switch functions such as hold, transfer, and call screening.
Users are identified to a SIP system by location-independent, email-like names; e.g., zippy@home.com. And like a browser requesting a home page from a web server, a SIP client sends requests to a receiving server, which processes them and returns proper responses. The SIP messages themselves are very simple, such as "INVITE" (invites another SIP user to a session) and "OPTIONS" (asks a SIP server to reply with its feature set).
SIP has kept up with the times. With the rise of Instant Messaging and Presence (IMP) services, another IETF working group developed an appropriate set of signaling extensions, called SIMPLE (SIP for Instant Messaging and Presence Leveraging Extensions). SIMPLE is practically a standard, but hasn't yet been finalized as one. Work is also underway to ensure that SIP will function with Mobile IP devices SIP's registration capability lends itself to provide mobility and location functions. Indeed, SIP will be used as the official 3G wireless multimedia protocol, as standardized in the Third-Generation Partnership Project (3GPP).
"Three years ago we founded the SIP Forum in part to show that there was industry support for SIP technology, but nowadays that's not an issue," says Jorgen Bjorkner, VP of concept development at Hotsip (www.hotsip.com) and chairman of the SIP Forum. "Vendors are very eager to put SIP on their products' datasheets and no one talks about H.323 anymore. The question now is really over which delivery model will ultimately dominate: Will it be the customer premise model, or will SIP services be delivered by the classic service provider-customer model?"
HotSip is betting on the latter. It builds SIP servers and SIP infrastructure for ISPs, telcos and service providers. These in turn offer services to end users who are typically residential but sometimes are corporate workers. HotSip doesn't build its own voice gateways for service providers, but it works with third parties. HotSip developed its own SIP softclient, called Active Contact, which is essentially a softphone with SIP, instant messaging, presence, and data conferencing support that interoperates with other SIP client software out on the network.
Noted popular SIP applications delivered by HotSip systems to end users are telephony and conferencing.
"There have been three major categories of SIP deployment," says Bjorkner. "First you have the background, infrastructure functions not visible to the end user. Second, there's the IP feature phone kind of applications offered by service providers, where you can use your PC or Pocket PC to place calls. Finally, there's redeployment, where a user with the proper equipment acquires a URL for a universal numbering system such as Electronic Numbering (ENUM), which can be used to both dial out and receive calls."
"Some companies prefer hosted services," says Bjorkner. "It's not obvious that the telcos of today will provide such SIP-based services, however. It could be some competitive carrier or ISP. This is the market we are addressing."
Ubiquity Software (www.ubiquity.net) also develops and markets SIP-based communications software for service providers, Independent Software Vendors (ISVs) and OEM partners worldwide.
Jeff Liebl is Ubiquity's vice president of worldwide marketing and product management. He says: "In the late 1990s we identified the SIP protocol as a very compelling proposition for solving some of the shortcomings of CTI and middleware and various protocols. We got really excited about SIP and turned the company in that direction. We built and demonstrated one of the very first SIP proxy servers in 1999. We left our old legacy business behind, and we now offer two major products."
One of these is Ubiquity's flagship product, the Ubiquity SIP Application Server, which is designed for carriers as a development platform for building SIP-based real-time communications apps. It also serves as a deployment platform for hosting multiple applications in a wireless line and mobile IP Multimedia System (IMS) framework for 3G wireless. It does load-balancing, can be clustered, and has a lot of high availability features.
The second major product is Ubiquity's Speak Conference Director, an application that's built and runs atop the app server. "It's a bridge media IP conferencing application," says Liebl, "and it uses the building blocks and the app creation environment of our SIP app server."
Ubiquity also has an application partners program. "We've recruited third parties to build SIP-based applications and migrate them onto our platform," says Liebl. "This would include so-called infotainment services. One such application that Siemens wrote on our platform is called VD-Infochannel, which Siemens managers use to supply their distribution staff with the latest distribution releases, product information and price lists."
"If you boiled our software down to the very basic essence, it's a very high performance SIP stack with a Java High Availability infrastructure," says Liebl. "We've built and incorporated a bunch of application building blocks within the SIP servlet container, things like Session Control, CPL, conferencing, IM, presence management and IVR. They're a bunch of building blocks that are exposed through our SDK that allow you to build very quickly applications without having to know a lot about SIP itself. You can add more building blocks into the container using that SIP servlet API on the other end."
One new application that excites Liebl is Push-To-Talk (PTT), a direct communications app reminiscent of a giant walkie-talkie system, since it can do one-to-one and one-to-many, half-duplex calls in an IP network setting. "We think push-to-talk is one of those great apps, considering the well-documented success that Nextel has had with it," says Liebl. "Many vendors are trying to do that in IP now, and we think that our partner Togabi Technologies (www.togabi.com) has got one of the best PTT solutions, called PacketCHAT. You can go anywhere in the US with a CDMA or GPRS service and you can demonstrate it with a one-second call latency time. It's pretty powerful, and it's all done with SIP using that always-on IP connection in the 2.5G environment."
Both Togabi and Samsung (www.samsung.com) announced recently that Togabi's standards-based PacketCHAT client technology is being integrated into a new generation of PTT-enabled mobile terminals built by Samsung.
"SIP will play an important role in mobile communications, whether it's in WiFi or 2.5G or the 3G market that's now emerging faster in Europe and Asia than in North America," says Liebl. "As IP becomes ubiquitous, you'll have some kind of device that's always registered in the network, and this opens up a world of more interesting real-time communications. And SIP will be there."
"Most analysts think that, by 2002-2008 or 2008, SIP-based VoIP usage will be evenly split between hosted IP services and actual IP PBXs. Of course, we'd like to see that happen sooner," says Steve Bakke CTO and co-founder of VocalData (www.vocaldata.com).
VocalData develops and markets "VOISS", a suite of IP voice applications and services for service providers and system integrators, who provide hosted PBX or IP Centrex services to business and residential customers VocalData's product is essentially a hosted PBX. "One of our claims to fame is that it's 100% carrier grade, has a 100% redundant architecture, and it can scale to over 100,000 users," says Bakke. "Our system can serve one business having 100,000 users, or 10,000 businesses, each having 10 users. We do a complete suite of applications along with the traditional hosted PBX functions. Service providers can support unified messaging, conferencing and call center functions."
"On a small scale," says Bakke, "you can run our software on one set of high availability, redundant Sun servers. There are, of course, media servers and PSTN gateway services to contend with, but in a small configuration, all of those services will run in a single box. As you scale up to 100,000+ users, the system becomes a series of pairs of boxes: one pair for call processing, another for the database, and another pair running web servers for the web portal. A TCP/IP network connects the boxes but they actually communicate with each other in the background via a protocol called the Open Application Telephony Specification (OATS). OATS is similar to SIP, but predates it. To the SIP user and SIP equipment, we're a SIP proxy server. Everything registers with us."
VocalData's customers claim a roughly 23% monthly savings by using a hosted PBX service. Major savings result from reduced line charges and the fact that packet voice doesn't require additional lines to handle call overflows. Voicemail is included in basic line charges.
Zultys Technologies (www.zultys.com) makes several interesting SIP-related items for the customer premise. One is currently the highest density phone system in the world, the MX1200 Enterprise Media Exchange, a 2U (3.5-inch) high, 19-inch wide rackmount into which is somehow crammed the functionality of a PBX, voice mail server, switch router, and Internet gateway. It even supports a boatload of applications, such as instant messaging, presence, and voicemail management. Customers can license capability for only the number of users they need to serve. The full capacity and/or capabilities of the MX1200 can be unlocked via software. The box is powered by four IBM PowerPC processors and relies upon SIP, TAPI, VoiceXML, and Linux. "We build products that offer high levels of integration," says Iain Milnes, president of Zultys. "By achieving such levels of integration we also have added collaborative features, such as SIMPLE-based presence and instant messaging, and we tie those collaborative features to the actual phone so that you can set the phone call handling rules based upon your presence at a particular location, for example."
Zultys is not supporting H.323 or any other similar protocol. "SIP is, in our opinion, the only way to go," says Milnes. "We guarantee interoperability of our products with the Cisco 7960 phone, as well as Microsoft Windows Messenger."
In June, Zultys announced the new ZIP 2 IP phone, a SIP phone that's affordable yet can meet business telephony needs. The ZIP 2 doesn't have a display, but supports many PBX functions, provides two simultaneous call appearances and supports three-way conference calling. The ZIP 2 is compatible with Windows Messenger and SIP-based IP telephony systems. A software application allows it to support up to 16 call appearances. The phone "binds" to the app through the Media Exchange. The same functionality can be attained using the Cisco phone or Windows Messenger. Zultys also offers the ZIP 4x4 phone. Launched in March, it's said to be the only phone available with a line rate switch and four external 100 Mbps Ethernet ports (you could plug your laptop or daisy chain other phones into them). It also has four call appearances, does call conferencing, call park, hold, encryption, and has advanced features such as standards-based IM that appears on the display.
The phones support both G.711 and G.729 voice codecs. Patrick Ferriter, Zultys' director of product marketing, warns: "A lot of Asian vendors are propagating really low cost SIP-endpoints. Many service providers and others snapped them up and tried to use them. Well, you get what you pay for. That's true across the industry regardless of the equipment. If you're getting an endpoint for less than half the cost of anything you've ever seen before, it probably won't work very well." As with any other kind of equipment purchase, one should always have some familiarity with a vendor and its interoperability lists before you purchase a SIP phone or a whole system, for that matter.
To sum up, SIP will be the dominant call control protocol. As long as IP phone vendors adhere to the standard and submit their products to international SIP testing events and "bake-offs", the interoperability of IP communications equipment will be assured. So start "SIP-ing" today!
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