Lippis Report Issue 89: Session Initiation Protocol (SIP): An Industry Update
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For many business and IT leaders Session Initiation Protocol (SIP) means cheap, low cost IP phones. And while there are low cost SIP phones available, holding on to that concept only allows you to miss the main point, which is that SIP is an ecosystem and fundamental to a new era in communications. True, the IETF’s SIP goal was to create a protocol for setting up and tearing down real time sessions over IP packet networks. But SIP has transcended that goal by emerging as a framework which engineers leverage to architect IP communication solutions. There are three aspects to the SIP ecosystem: SIP connections, SIP end-points and the SIP platform. In this Lippis Report we explore the SIP ecosystem as the underpinning of unified communications.
The SIP ecosystem:
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There are three aspects of SIP. First SIP connections which consist of SIP trunking, gateways and peering offered by various service providers and equipment suppliers. There are a wide range of SIP end-points including soft-phones, hard-phones, fixed and mobile end-points which make up the second component of the ecosystem. Linking SIP connections to end-points requires a SIP platform which is a switching mechanism establishing and disconnecting sessions and offering application developers hooks to create and customize communication applications. In short, the communications industry has divided itself into these three areas, making up the SIP ecosystem.
Connection Points: SIP Trunking and Peering
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The connection segment of the SIP ecosystem provides links into service providers via SIP trunking and between service providers via SIP peering. SIP trunking is being provided by many of the equipment suppliers such as Avaya, Cisco, Siemens, Nortel, Alcatel, ShoreTel, Mitel, et al. natively without the use of separate gateway device or appliance. The value proposition of SIP trunking is one of lower communication cost and simplicity thanks to its being able to provide a service provider with a single SIP link (T1, DS3, etc.) A single SIP trunk provides the transport of many simulation SIP sessions vs. the alternative of linking buildings with private lines to support SIP traffic or overloading existing enterprise WANs and routers with backhauling inter-company SIP traffic. SIP trunking simplifies numbering too. In short SIP trunking delivers traffic aggregation, addressing, and naming complexity reduction, which lowers operational and facilities cost.
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The number of service providers offering or announcing SIP trunking has grown considerably over the past 24 months. In North America SIP Trunking is either available or soon will be from AGN Networks, AT&T, BandTel, Global Crossing, Onvoy, Paetec, Verizon Business, McleodUSA, Qwest, and TelePacific. In EMEA (Europe, the Middle East and Africa) SIP trunking and peering is being provided by Arcor, Belgacom, British Telecom, Colt, Deutsche Telekom, Proximus and QSC. CALA (Central America/Latin America) SIP service providers include Embratel, Telefonica and UNE - EPM Telecommunications. In Asia and Pacific (APAC) KT (Korean Telecom), SingTel, TFN (Taiwan Fixed Network), and TOT (Telephone Organization of Thailand) are SIP service providers. This level of global SIP support is a testament to the interoperability that SIP provides. For example, SIP end-points are different across the world thanks to the multitude of suppliers yet all can connect and communicate via SIP trunks and peering points between the above SIP service providers.
SIP end-points
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SIP end-points are not about cheap phones. SIP offers end-point value well beyond a price point. Perhaps the most prevalent SIP end-point will be the soft-phones available from Microsoft and IBM as both use SIP as the cornerstone to their unified communications products.
SIP is opening up end-points for customization, greater user control, branding, etc. SIP allows extending features and functions between hard-phones, soft-phones and mobile phones. In some SIP product lines such as the Avaya one-X, users enjoy the same user interface independent of hard-phone, soft-phone or mobile phone, providing consistency, ease of use and some very cool features such as call logs, especially missed calls, and twinning of devices. Twinning devices rings both desk and mobile phones so that if you’re not at your desk, but you’re walking around the building or outside SIP is ringing both your phones.
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Whisper paging is another cool feature that’s becoming very popular and in demand. Whisper paging allows an assistant to break into a boss’s call with only the assistant and boss hearing each other. In essence the assistant is able to whisper a message into the boss’s ear and ask if they’d like to take the call. These are the types of features that are traditionally available in the PBX space, but completely non-traditional on mobile devices. What SIP has permitted is to extend some of those features and functions all the way out to mobile devices. In addition to these feature examples, many IP telephony providers are adding features such as web browsing to their SIP phones, which enable customization and new services.
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True, the IETF SIP standard defined a limited number of features such as place call, forward call, three-way local conferencing, put call on hold, transfer call, etc. And many suppliers are extending the feature set through SIP extensions, which can be a double-edged sword. On one side SIP extensions increase the feature set, on the other side some suppliers may build proprietary extensions which operate only with their SIP platform, locking customers into their architecture for years to come.
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While most, if not all, SIP end-point providers deliver on basic features, where differentiation occurs is in the extension of traditional PBX features to SIP phones. For example, not all SIP phone providers are able to provide bridge line appearances because there is no SIP RFC standard. Bridge line appearance allows an administrative assistant to know if the people he/she supports are available to take a call. There is a draft, which describes how to do bridge line appearances or the busy indicator of somebody else’s appearance. In SIP the leveraging of presence can deliver a wide range of older PBX features and as well as new ones. There is a method to extend SIP that is compliant to the SIP standard as well; the spirit of SIP using presence to provide bridge line appearances is an example of that method. Business and IT leaders should select SIP vendors carefully and choose those who implement SIP per IETF draft and the RFCs.
The bottom line: SIP end-points are not cheap phones but are becoming feature rich communicating devices in many form factors and price points.
The SIP platform:
While SIP end-points get most of the press attention, the SIP platform receives the lion’s share of business and IT leader attention. Large corporate IT staffs are spending approximately 25% on SIP end-points and 75% on the SIP platform. Not only does the SIP platform demand the lion’s share of acquisition dollars but a lot of time is being spent on planning how the SIP platform will fit into an IT infrastructure.
Most of the hard SIP work was in the establishment of the protocols and framework, which defines the SIP platform. The industry has looked at the two edges of the SIP ecosystem (end-points and connections) but not enough on the platform in the middle. In order for the two edges to work end-to-end you need the middle. The SIP platform provides session establishment and management, a presence server, a registrar and the platform proxy. Beyond the basic session services the SIP platform is also providing a development environment, which allows IT developers to either customize specific communication applications or call upon SIP services to add value to structured business processes. In the real world the SIP platform will be the evolution of Avaya’s communication manager and its ubiquity application development platform. For Cisco it would be the evolution of call manager with its Reactivity acquisition. Perhaps the best example of this is the Ubiquity SIP A/S, which is the SIP Application Server. There is a whole developer network that has been formed around SIP A/S resulting in a wide range of applications.
A New Era In Communications
The SIP ecosystem is changing the communications and computing industry. With Microsoft and IBM entering into the market, it’s clear that the days of the $600 desktop phone are limited. The reason? Microsoft and IBM will give back to IP telephony providers approximately $8 per soft-phone license. In essence Microsoft and IBM are gearing up to be the new distribution channel for enterprise communications. The revenue loss due to the dwindling of the $600 per fixed phone will have two results to the enterprise IP telephony providers. First there will be consolidation and fewer IP telephony providers over this next business cycle as there will be less revenue to support the current number of vendors. Second, those IP telephony providers who embrace a software and services model will be better equipped to navigate the industry change. While soft-phones and mobile/PDA phones will grow at the expense of fixed desktop phones, those IP telephony firms who add value to their soft- and PDA platforms will be better positioned to increase margin well beyond what their desktop phone revenue stream provided.
Expect all IP telephony providers to compete aggressively on unified communications with feature richness and SIP platform robustness. Adding “chargeable” value to PDA and mobile phones will be key to success. Bolstering their SIP platform to go beyond basic SIP services to include an application development environment will be key too as this is where professional services and customized solutions reside. We have entered a new era of communications, one that is marked by openness, software and integration and SIP at the center of it.

sflau01 said:
September 13th, 2007 at 8:11 am
Hi,
I have difficulty downloading some whitepapers identified on the right hand side of this document. Ths download seems OK but an error prevents it to be displayed. Is there a way to rectify this problem. The Whitepapers that I am interested in are:
1. Session Border ControllerDelivering Interactive Communications Across IP Network Borders
2. SIP Trunking Benefits and Best Practices
3. Integrating Telephony Services Into .NET Applications
Thanks,
Stephen Lau
VoIP, Led by SIP, Forges Ahead - Caller IP said:
November 25th, 2007 at 8:28 am
[...] The folks at the Lippis Group do their usual comprehensive job in this report on SIP. Indeed, the piece has a double benefit. On one hand, it breaks overall SIP technology into three easier-to-understand elements: SIP connections, SIP endpoints and the SIP platform. The piece defines each of these elements and, perhaps most importantly, lists the main players in each. For instance, the key vendors in the SIP connections category are Avaya, Cisco, Siemens, Nortel, Alcatel, ShoreTel and Mitel. The piece says that many service providers have entered the trunking segment during the past two years. In the U. S., new service providers include AGN Networks, AT&T, BandTel, Global Crossing, Onvoy, Paetec, Verizon Business, McLeodUSA, Qwest and TelePacific. [...]