Open Source — software where the source code is available for anyone to improve or modify —has the potential to change the telecom landscape forever. Already Linux has made huge headway into this market, and the advent of Open Source has caused quite a stir. Developers need to know what is available in the Open Source marketplace and how best to leverage the freedom inherent in a distributed community of programmers to amend and adapt code for their VoIP development needs. This session will address the role of the various emerging VoIP standards and what they mean to the development of Open Source.

You continually hear about SIP, but do you find yourself bewildered by the buzzwords and terminology? Wish someone would tell you what a SIP Proxy was, and why it might feel the need to fork? What advantages does SIP bring? Why should I move now? This ‘back to basics’ session will take a high level look at the SIP protocol and the power it puts in the hands of application developers and solution architects. The presenters will discuss the building blocks required to put together a SIP deployment so you can talk with confidence about how the next generation of telephony products will be built!

SIP is a promising protocol in VoIP due to its simplicity and flexibility. For better or worse, the barriers for developing SIP-compliant devices have dropped significantly. You can find SIP products from developers who work out of their garage. For carriers who operate SIP-based networks that interoperate with this garage-baked equipment, it is crucial that they can compensate and improve the speech quality from the low-budget equipment so that the overall voice experience does not degrade. This presentation examines these SIP interoperability issues in detail and provides practical solutions for carriers who are facing speech quality issues in their networks.

As service providers begin real life implementations of SIP networks and products, they are faced with numerous issues including interoperability and quality of service to name just a few. In this session, the speaker will illustrate with examples things that work and areas that still require significant industry effort. In addition, this presentation will identify and address the challenges that vendors and service providers face as they deploy SIP networks.

There are several IETF RFCs and Internet Drafts that describe mechanisms for securing SIP. Many of these mechanisms are difficult to understand and implement. The reSIProcate project at SIPfoundry has implemented several of these schemes to provide a complete SIP-based security system including end-to-end security using S/MIME and hop-by-hop security using TLS. It also provides an implementation of a mechanism to distribute and store public and private keys using SIP. This presentation will describe the capabilities of the reSIProcate libraries and how companies can take advantage of them in commercial products.

As support for SIP continues to grow, creation of SIP-based services becomes a key element for developers to focus on as they scramble for market leadership. This session will provide a close look at examples of promising enterprise, wireless, and wireline applications of SIP, including Push-to-Talk, Infotainment, and the like. The speaker will also focus on the promising wireless applications enabled by SIP as well as the technology and market challenges facing the SIP community.

SIP and the applications it supports, from management tools and interactive meetings to voice-enriched e-commerce, are continuously reshaping the way today’s organization conducts business, by creating increased productivity and enhancing efficiency. This session will address the ins and outs of SIP: how it operates, the advancement of SIP-based solutions, the different diagnostic and support tools that have enabled this protocol to gain such wide acceptance, best practices for deploying SIP solutions, and what the future of innovative SIP-enabled VoIP applications holds. The extension of SIP to provide innovative services and accommodate the requirements of real world deployment, where NATs, service level agreements and regulators exist is also covered.

Gartner predicts that 90 percent of all new corporate telephone networks will be IP-enabled and based on SIP protocols by 2008. Still, concerns remain around the security of VoIP and the underlying SIP protocol, fearing that they are susceptible to similar types of threats and exploits that plague the Web and e-mail. Today organizations of all sizes need to evaluate and understand the security measures available that allow companies to deploy real-time messaging, voice, data, video and other SIP based applications with confidence. In this session, the speaker will examine specific SIP security problems including voicemail spam, identity theft, impersonation, session eavesdropping, voicemail bombing, hijacking and redirection. Topics will also include the steps that need to be taken to secure and manage the dynamic nature of real-time SIP communications.

Real-time person-to-person communications are fast becoming a critical communications tool for enterprises of all sizes. With the standardization of SIP as the Internet protocol for applications such as VoIP, instant messaging, presence, and video, businesses are eager to adapt their existing hardware to accept SIP quickly, cost-effectively and securely. Traversing the firewall is a tricky proposition when integrating SIP into any enterprise. This session will cover the basics of SIP, its history, evolution, and predictions for the future, with a focus on solutions for traversing the firewall for SIP-based communications.

This session will take a look at how SIP will dramatically change ingrained enterprise applications such as CRM, ERP, and contact centers. SIP's inherent multimedia capabilities will provide innumerable opportunities to greatly improve productivity, customer interaction, and user experience. Examples will be provided in this exciting look to the future of SIP enabled applications.

ENUM is considered by many to be the key ingredient for the successful convergence between PSTN and Internet. ENUM enables the routing of telephone numbers to IP thus enabling a more efficient method of routing VoIP calls to VoIP recipients (bypassing the PSTN altogether) and also facilitates PSTN to VoIP calling. SIP continues to evolve and is now an ever-present standard in worldwide VoIP deployments. Come to this session to learn all about ENUM and SIP: what they do, how they interrelate, as well as other considerations in these types of deployments.