Does moving the media processing function from DSP to the host fundamentally change the architecture? Yes! Host Media Processing (HMP) offers new opportunities for voice-enabled applications to be architected — to create a very broad range of scalability; to partition functionality in very different ways; and to create highly reliable services. This session will also discuss new ways to process media that take greater advantage of the host environment.

VoIP development platforms are a constantly evolving lot. From PCI and CompactPCI to today’s most advanced ATCA and microTCA-based platforms, there are so many changes that it might be tough to stay abreast of all that is happening in this fast-moving space. Attend this session, and you’ll hear the presenters talk about powerful, standards-based computing solutions and hot swap and high availability capabilities for communications applications and how it all relates to your development plans.

DSPs enable equipment manufacturers — and their carrier customers — to launch secure, high-quality, scalable broadband services over IP. Such processors reduce the development time and cost associated with delivering a range of broadband services, beginning with VoIP and extending to full triple-play voice, video, and data offerings. The session will pay special attention to triple play application development and utilizing advanced DSP algorithms to remove noise impairments while enhancing the intelligibility of speech in environments with high ambient noise levels.

Wideband speech provides a key quality differentiator from other VoIP offerings and from traditional telephony by more than doubling the encoded signal bandwidth and providing not only richer sound but much better intelligibility that improves communication efficacy and helps speaker recognition. It provides a sense of presence for applications like videoconferencing and teleconferencing as well as any conversation or broadcast. This session focuses on the market drivers, advantages and characteristics of wideband VoIP telephony. In particular, it provides an overview of the key features and functionality of the highly efficient G.722.2/AMR-WB/VMR-WB family of wideband codecs.

Security remains a concern in packet networks. This session will address a number of issues regarding development of VoIP security solutions, including encryption, user authentication, denial of service (DOS) attacks, and the multiple network layers (signaling, media, transport) subject to security testing. The presenters will also look at semiconductor technology, including system-on-a-chip (SoC) architecture and voice processing software, which provides a platform for cost-effectively deploying VoIP services in support of the latest VoIP security technologies including PacketCable Secure Voice, and Voice Over IPSec.

While VoIP has many positive benefits to corporations, the potential security risks involved should not be ignored and must be effectively addressed before a loss in money or private data occurs. New products and services must be created to eliminate these security threats. The speakers will address the growing need for the communications industry to address and develop VoIP security solutions now and for the future. The session will cover how implementing security at the “core” of the VoIP system, the platform level, and why ensuring that encryption is synched from the device and network is critical.

As the telecommunications industry evolves to embrace IMS, the delivery of unique service combinations requires a horizontal infrastructure that maximizes reusability and minimizes service delivery costs. This session will explore how service providers can develop next generation network convergence architectures that leverage IP and SIP based technologies to bridge mobile and fixed networks while preparing for the future of IMS.

As the telecom industry increasingly turns to offer new services such as IPTV and Triple Play bundles, developers will increasingly be called upon to enhance existing services and create brand-new revenue generating applications to drive a successful bottom line. The product chain extends from the core of the service provider’s network out through the set-top box hat sits in the customer prem. This presents a tremendous opportunity for developers. Come to this session and hear what tools are available and what you need to know to develop IPTV applications.

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.

This group discussion will zero in on the current state of Open Source telephony; what it means, where it stands, and where this segment of the market is headed. Thought leaders from a number of influential Open Source factions will share their insights with the audience. Attendees can expect a frank conversation, and will leave this session with a thorough understanding of Open Source and its role in the future of IP Telephony.

As with any technology element, subsequent product generations typically yield performance, size, integration, or cost advantages. This is often driven by the silicon devices used to define the product. Key to leveraging the advances in silicon has been the equal advances in embedded software. A leading element is the ongoing growth and maturity of Linux as a fully functional and flexible operating system environment, as well as a configurable environment for embedded systems. The performance and integrated benefits of contemporary silicon devices coupled with the functionality and cost benefits of Linux are providing embedded developers with options never before available. This session will discuss the use of Linux as an embedded operating system and development environment and how it can be applied towards the development of embedded systems.

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.

Imagine opening night for a play. The actors are in costume and know their lines, but they have not yet rehearsed together. Unthinkable as the above scenario might be, a very similar risk is in the offing for carriers preparing to launch IP telephony services without testing the performance and security of their VoIP networks end-to-end, before raising the curtain to a paying audience. This session will examine the potentially fatal difference between testing VoIP network components and testing real world VoIP services under real world, end-to-end network conditions. Topics will include: emulating real-world traffic from multiple points on the network; test the full range of IP communications and collaboration applications likely to travel the network; and the full range of network statistics, call measurements statistics, and voice quality metrics.

Are wireless networks and technologies mature enough to deliver voice? It’s certainly easier to successfully deliver data than voice. Data can be rapidly transmitted without worrying about the effect of network delay, latency, QoS, etc. WLAN providers, equipment manufacturers, network designers and operators must now prove that WLANs and wireless handsets can meet the strict performance requirements required for high-quality delivery of voice. To achieve acceptable voice quality, they must understand relevant performance requirements that affect the network’s ability to manage voice and other real-time applications. This presentation will discuss performance requirements and the emerging standards that are being developed to support voice.

This session will discuss how VoIP and VoWiFi solve real problems for enterprises and consumers alike. An “inside-view” of VoWiFi phones and the next generation of dual-mode phones (cellular + VoWiFi) will be provided, with a discussion of the key issues that enable, as well as hinder the success of this convergence platform. Topics to be discussed include developer’s view of VoWiFi and next-gen Dual-mode phones; benefits of VoIP enabled convergence for mobile communications; and key issues hindering rapid adoption of dual-mode handsets.