At the heart of the pending freedom is a protocol still in development by the Internet Engineering Task Force, or IETF, a consortium of computer scientists and programmers that creates and establishes universal standards for the Net. Called "SIP for Instant Messaging and Presence Leveraging Extensions," or more succinctly, SIMPLE, the forthcoming IM standard will establish a uniform means for short text-messages sent over the Net. With a common standard in place, IMs could easily travel between different SIMPLE networks - just like calls can be carried between Sprint and AT&T phone networks. Moreover, SIMPLE uses another IETF standard called Session Initiation Protocol, or SIP, which was designed to handle much more complex Net traffic such as digital video and voice. Next generation IMs that use SIMPLE would then be able to handle digital movie files or even telephone-like conversations over a single simple interface.
The monolithic circuit switches of the traditional PSTN are well known. There are over 19,000 Class 5 switches deployed in central offices and over 500 Class 4 tandem switches deployed in regional offices across the United States alone. While the annual growth rate of these voice switches has recently been slow and steady, the traditional voice switching architecture will soon undergo an evolution similar to that of the computing architecture in the early 1980s, in which advanced PC workstations running third-party software replaced vertically integrated proprietary mainframes. --- SIP (Session Initiation Protocol) is an IETF protocol for transporting call control, authentication and other signaling messages among softswitches and other devices through an IP network.
As we have seen, softswitches provide compelling value for streamlined communication networks. SIP provides an evolution of communications into peer-to-peer multimedia. While dedicated SIP application servers are rolling out under their own banner, we also see softswitches being positioned as SIP servers. The de-evolution of the switch is spawning increasingly specialized servers, and SIP is likely to accelerate that trend. In addition, SIP's ability to span different network types will help further the differentiation of SIP-enabled softswitches dedicated to different markets. Together and apart, softswitches and SIP will continue to accelerate the transformation to the next-generation network.
The worlds of telephony and the Internet are converging. If your reaction to this statement is, "Tell me something that I don't know," you may want to think about it a little bit more. Do you really know how this convergence will affect your business? Are you fully aware of the opportunities it will bring? The worlds of telephony and computing are changing. And they are changing fast. It started with telephony's increased reliance on Internet Protocol (IP) as the telecom network technology of choice, and it has continued with a migration toward software-based telephony networks that incorporate software-based switches and standard protocols such as Session Initiation Protocol (SIP) that are modeled heavily after Internet protocols such as HTTP.
Session Initiation Protocol (SIP) is quickly gaining popularity with application service providers (ASPs), communication service providers (CSPs), and network service providers (NSPs) focused on offering their customers innovative, new IP-based services. Adopted in 1999 by the Internet Engineering Task Force (IETF), SIP provides for the seamless transmission of voice, fax, and data across IP and traditional telephone networks. The IETF defines SIP as "a text-based protocol, similar to HTTP and SMTP, for initiating interactive communication sessions between users. Such sessions include voice, video, chat, interactive games, and virtual reality." SIP is used to establish peer-to-peer media sessions on an IP network including Internet telephony, conferencing, instant messaging, and unified messaging. It uses less bandwidth than H.323 because it requires fewer and smaller messages to set up and tear down a call. SIP has become a standard IETF protocol for signaling in third-generation mobile networking because it is able to initiate, modify, and terminate calls on the network and because it allows end users to become "terminal independent."
In the first public test of Session Initiation Protocol - the challenger to the more established H.323 standard for voice-over-IP call control - vendors proved they could achieve basic interoperability with minimal tweaking.
An emerging communications protocol called SIMPLE is the front-runner to become the standard method for sharing online presence information and instant messages across the Internet, thanks to backing from market leaders AOL Time Warner and Microsoft.
Taiwan's state-owned telecom operator Chunghwa Telecom, of Taipei, is poised to become the first regional carrier to deploy a new generation of Internet services based on the telephone. Session initiation protocol (SIP)-based services have been launched by operators in Europe, but Chunghwa will be the first to roll out commercially in Pacific Asia, according to sources close to the company.
A year ago, we said SIP would win the “protocol wars.” Today, however, SIP's biggest challenge comes not from rival protocols, but from a timid market's hesitation to explore the stunning new application paradigms that it enables.
"I'm hearing that NetMeeting will not see any new functionality, though, and the team has been folded into MSN Messenger and any new functionality will be a part of MSN Messenger in the future," said Robert Scoble, founder of the DevX NetMeeting Zone. "Basically I've been told that 'NetMeeting is dead and everything is getting rolled into MSN Messenger,'" Scoble continued.See also "Microsoft Is Ready to Supply a Phone in Every Computer", New York Times, June 12, 2001.
To ensure that Windows Messenger has mass appeal, Microsoft has made the product compliant with the Session Initiation Protocol (SIP) advocated by the Internet Engineering Task Force. SIP is a framework for establishing, maintaining and ending Internet multimedia conferences and phone calls.
We've all heard the hype about Voice over IP (VoIP) technologies, and their promise of cool new services-services so revolutionary that we can't even dream of them. Right. As anybody knows, VoIP constitutes a blip on the radar of voice traffic, and networkers would be hard-pressed to point to any compelling reason for going with VoIP today.
So why then would Network Magazine give a Product of the Year award to the Session Initiation Protocol (SIP), yet another VoIP protocol, this time from the IETF? Quite simply, it's because SIP isn't just about voice. It's the protocol that will provide the foundation for all integrated communications-sessions combining voice, fax, data, and video-over IP networks in an easy, cost-effective way.
Those last two points are critical. Until now, developers and service providers needed to use the H.323 protocol from the ITU, which has come under fire on two fronts. For starters, H.323 and Web developers don't really mix. As Jeff Skelton, chief technical officer at Net2Phone, a leading VoIP carrier, noted in our February 2001 story on SIP ("Calling All Carriers"), "It's harder for developers to pick up and adopt H.323 because of the binary message format." SIP's HTTP-like structure avoids those problems.
Then there's the sharp criticism on H.323's implementation costs. "We could have implemented H.323 in our IP telephones," says Ed Wadbrook, director of strategy in application development at 3com, "but that would have meant spending $75 to $100 [per phone] on extra memory and hardware."
SIP pundits are betting that the aforementioned points will pave the way for carriers to finally roll out the innovative voice services only possible with IP. And just what will those services deliver?
Lots of Web integration to simplify follow-me services, call conferencing, and ways for users to speak with a live agent just by clicking a Web site button. SIP also standardizes presence, the generic term for Instant Messaging (IM), bringing a whole new angle to VoIP, such as establishing conference calls on the fly when all participants are online.
Indeed, a number of carriers-including Level3 Communi-cations and WorldCom within the United States, Telia in Sweden, and VoIP operators DeltaThree and Net2Phone-are testing SIP. But those creative services will only be possible once SIP application creation platforms ship in earnest this spring. Then, watch out: Cool services, lower implementation costs, and easy-to-use interfaces will make SIP the hot ticket of communications protocols in 2001.
Power to the telephone masses
A protocol that allows voice, data, fax, video, instant messaging and even online gaming to be integrated with web-based applications
Think of what the PC did to mainframe computing and the makers of "heavy iron". That gives some idea of what the SIP phone could do to the traditional telephone system and the telcos that operate it. SIP is to VoIP (see article) what the PC was to the radical notion two decades ago of distributed processing-in short, the user-friendly gizmo that ushered in a wholly new way of doing things. So with SIP.
The Session Initiation Protocol (SIP) is emerging as the favoured standard for setting up, modifying and terminating telephone calls over the Internet. Its main attraction is that it puts all the power of the network-plus the ability to change things on the fly-in the hands of the user.
Want to set up instant telephone conferencing, voicegrams, follow-me, a global phone number, text-to-speech delivery or online statements? No more waiting months for the phone company to program such features (if actually available) into its local telephone exchange. With the click of a button or mouse, SIP phone users will be setting up all these things and more for themselves the instant they need them-and cancelling them, if they choose, the moment they have finished with them. Press a couple of keys and all phone calls to your daughter are blocked while she finishes her homework. Hit a couple more and the bar is removed and her accumulated voicemail played back.
Giving such control to the end-user is anathema to the telephone clergy. As befits an organisation that oversees a vertically integrated global network with control concentrated at the centre, the International Telecommunication Union would have all VoIP suppliers adopt a Byzantine standard called H.323. This is derived from an old video-conferencing protocol that specifies everything but the colour of the knobs and switches. There is no question that H.323 works, but it is needlessly complicated and understood only by the anointed few.
By comparison, SIP could not be simpler, being modelled on the Hypertext Transfer Protocol (HTTP) used for specifying web-pages. Unlike H.323, SIP does not try to specify anything it does not have to. Instead of using arcane codes drawn from telephone signalling, SIP defines how a call should be set up, modified and torn down afterwards in the form of simple text commands-something which thousands of web-programmers can do blindfold.
But the really clever thing about SIP is that it makes a complete distinction between establishing a communication session between two or more parties, on the one hand, and what that session actually is on the other. That means SIP's controls for establishing, modifying or terminating a session can be applied equally to any kind of session-be it a telephone call, videoconference or multiplayer game of "Doom". In short, SIP is much more than just a smart telephone. It allows voice, data, fax, video, instant messaging and even online gaming to be integrated with web-based applications. The possibilities for e-commerce are endless.
And because it has its origins in the web, phone numbers on a SIP-based network become effectively the same as e-mail addresses-ie, email@example.com. A glimpse of this can already be seen in Japan. Subscribers to NTT Docomo's popular i-mode service use their phone number as part of their e-mail address (ie, firstname.lastname@example.org), allowing them to send and receive e-mail messages from their phones while chattering away. SIP is going to let users do even more-and free them, at the same time, from the slow and heavy hand of the telephone company's central office.
The adoption of intelligent end points in the enterprise is moving forward at a pace that is outstripping earlier expectations, and has thrown vendors and developers wanting to stay in the game into a breakneck race to get products to market. Using the signaling capabilities of the session initiation protocol (SIP), a new generation of smart phones, conferencing units and even soft phones for PCs will bring new power to desktops and pump even more energy into the already accelerating market for voice on the LAN. According to a new study from The Phillips Group InfoTech (www.phillips-infotech.com) titled "IP LAN Telephony: Market Demand and Implementation Strategies," U.S. companies are dumping their dual network infrastructure and traditional PBX systems in favor of single IP-based networks 30 percent faster than previously anticipated, leading the analyst group to predict that more than 80 percent of U.S. enterprises will have some form of IP LAN phone system in place within four years.
To find out just how these proxies fare against one another, Network Magazine compiled the first ever exhaustive analysis of SIP proxies. What's more, we've worked with Deltathree to provide a rating system for weighing the myriad features that distinguish these products. A summary of those results can be found in this table; see The Score on SIP Specs for a complete features listing and weighting of categories. Nearly 60 features were considered in the analysis of the technical capabilities of these critical servers.
WorldCom is using Cisco gateways to support its new offering, but says it will support other vendor gear in the future. The Cisco gateways support Session Initiation Protocol (SIP), which is used to send and receive voice calls between the PSTN and IP networks. SIP is a signaling protocol that sets up and tears down voice and data sessions over IP networks with less delay than H.323. SIP has gained a technical following for delivering nearly imperceptible voice-over-IP call setup delay compared with H.323, which was designed for heavier applications such as desktop video and data conferencing. SIP offers more flexibility and extensibility than H.323 for WorldCom and others to provide popular voice features over packet networks and develop new applications, says Fred Briggs, WorldCom's chief technology officer. SIP is a simpler protocol than H.323, says David Passmore, research director at The Burton Group. SIP uses XML and standard Domain Name System for IP address resolution. Users should have more SIP applications to choose from because application developers will likely find the protocol relatively easy to work with.
Last updated by Henning Schulzrinne