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file icon A Sip of SIP (IBM)hot!
17.11.2004
November, 2003

Abstract:   SIP (Session Initiation Protocol) is a standard protocol that covers a wide variety of applications. The document explains why SIP is a powerful real-time collaboration protocol. SIP is currently being adopted by many companies and is evolving more rapidly than any other existing real-time collaboration protocol. This document provides background information on SIP and touches the wide variety of applications whose implementation is either made easier or made possible, via SIP. This document concludes with a set of questions and answers on SIP.

Hits: 11869
file icon B2BUA: Enabling Class 5 Capabilities in SIP Designs (Radvision)hot!
17.09.2003
Enabling Class 5 Capabilities inSIP Design
Hits: 4253
file icon Big Communications Picture (Siemens)hot!
02.02.2004
This white paper provides an overview of the companyâ_Ts HiPath portfolio, the focus being on HiPath OpenScape, which is a suite of presence-aware, real-time communications software. HiPath OpenScape is a groundbreaking addition to the HiPath solutions portfolio and an important development in its own right. Additional Siemens white papers are available that focus on the portfolio, the HiPath MobileOffice solution and the companyâ_Ts award-winning migration strategy.
Hits: 3655
file icon Enterprising with SIP - A Technology Overview (Avaya)hot!
02.02.2004
The vision that Avaya has for converged voice and data communications is to create a network environment that supports maximum enterprise efficiency. Session Initiation Protocol (SIP) [18] is a key technology for this evolution towards a converged communication architecture. SIP as an enabler of converged communication applications has been able to gain industry-wide acceptance by placing control of the communication across distributed networked entities and by providing a means of integration at various levels in the protocol stack. Such integration is now possible in a multi-vendor environment with multiple applications across a wide range of industry sectors. Converged communications that is driven by ubiquitous IP network connectivity defines the next stage of evolution in enterprise communications.

SIP is an application layer, peer-to-peer communication protocol that facilitates openness, simplicity, flexibility, and reuse in an "All IP" Architecture. This paper offers a look at how Avaya views SIP in enterprise evolution. It also describes SIP technology, elaborates on the role of SIP in realizing converged communications, and describes details of the SIP protocol and associated technologies.

Hits: 6797
file icon Evolving to Converged communication with SIP (Avaya)hot!
11.05.2005
Organizations with multi-vendor, multi-application and widely distributed communications systems need a reliable way to integrate and optimize their network infrastructure. Session Initiation Protocol (SIP) supports converged communication applications by placing control of the communication across distributed networked entities and smoothly integrating the various system levels. SIP enables seamless data and voice convergence in multi-vendor environments with multiple applications across a wide range of industry sectors.

This white paper is designed to help business leaders understand the impact of SIP-based converged communication in Voice over IP (VoIP), Unified Communication and Customer Relationship Management (CRM) solutions. This paper also offers CIOs guidelines for migration to an enterprise SIP network, and details some new Converged Communication Solutions.

Hits: 7022
file icon Genesys IP Media eXchange Testing Reporthot!
11.09.2006
CT Labs was commissioned by Genesys to evaluate the call-handling performance of the IP Media eXchange (IPMX). The IPMX compo-nents were deployed on one, two, and three Windows-based servers, respectively, for each of the three tests described in this report. All VoIP calls were generated using an automation script running on the Empirix Hammer FX-IP platform, using the SIP protocol. In the first test, the IPMX was deployed on a single server. The results of this test reveal that a single high-performance server with 150 agents configured can effectively process in excess of 250 simultaneous callers at a rate of 8 calls/second, with half listening to music on hold, and the other half connected to agents.
Homepage: www.ct-labs.com
Hits: 2566
file icon Ingate Systems: SIP Trunking Benefits and Best Practices hot!
19.06.2007
SIP Trunking Benefits and Best Practices By Janne Magnusson, Vice President, Product Management Ingate® Systems A complete guide to SIP trunks, this whitepaper provides a comprehensive overview on the basics of SIP trunking, including a discussion of infrastructure needs and deployment issues and solutions. In-depth sections dedicated to interoperability, security and quality and reliability issues are also included. Editorials from ShoreTel, Level 3, Cbeyond/SIP Connect and Ingate provide a unique look at the perspectives from the IP-PBX, ITSP and other thought leader points-of-view. SIP trunks can lower telephony costs and deliver rapid return on investment (ROI). They also offer the opportunity for enhanced communications both within the enterprise and with vendors, customers and partners. While some see SIP as just voice, SIP trunking can also serve as the starting point for the entire breadth of realtime communications possible with the protocol, including Instant Messaging, presence applications, whiteboarding and application sharing. Adding their unique insights to the white paper are editorials by: - ShoreTel, Inc. – “Developing a Working Solution End-to-End” - Level 3 – “A Service Provider’s Perspective on SIP Trunking” - Ingate Systems – “Who Is Listening to My Calls?” - Cbeyond – “SIP Connect: An Industry Collaboration Success Story”
Homepage: http://www.ingate.com/files/white_paper_What_is_SIP_Trunking_A.pdf
Hits: 695
file icon Navigating the Complexities of IMS (Empirix)hot!
07.09.2006
IP Multimedia Subsystem (IMS) is moving the communications industry towards a new business model. No longer are proprietary solutions with long development cycles and network-centric applications the solution for the network equipment manufacturer or service provider community to enable feature-rich services. Traditionally, end user, network, testing/monitoring, and maintenance applications have been dictated by the access and network infrastructure equipment capabilities and feature sets. These applications have been integrated into the legacy equipment and the ability to customize has been extremely limited. However, these applications due to their proprietary and static nature and long development timeframes are very reliable and provide a high quality end user experience. This quality and reliability needs to be duplicated in the IMS environment and applied to real-time, next-generation applications but in a much quicker timeframe across diverse networks and devices.
Homepage: http://www.empirix.com/
Hits: 2262
file icon NexTone SBC Attack Performance Test (CT Labs)hot!
11.09.2006
CT Labs was commissioned by NexTone to verify SIP call-handling performance of the NexTone MSC session border controller (SBC) product while subjected to various real-world attacks. The test involved generating high levels of valid real-world SIP-based VoIP traffic while launching attacks against the SBC, including denial of service (DoS) and other malicious attacks designed to disrupt SIP-based services.
Homepage: www.ct-labs.com
Hits: 2169
file icon One Click Management for VoIPhot!
15.05.2007
Codima Toolbox delivers unrivalled pre assessment and post deployment VoIP testing and readiness for VoIP networks from beginning to end. This includes network pre assessment tools, traffic simulation, post deployment monitoring and troubleshooting combined with modules for Flow analyzing, RTP analyzing and Call Playback as well as inventory and visual mapping tools. Flexible and scalable Codima Toolbox is a set of tools easy to integrate, combine and scale ensuring successful VoIP implementations.
Homepage: http://www.codimatech.com
Hits: 590
file icon Open Source IP Communicationshot!
12.05.2005

SIPXchange, the industryâ_Ts first open source Enterprise communications suite, combines the cost savings of voice-over-IP technology with the quality and adaptability of open source software created by a community of developers; itâ_Ts a solution that can improve a business exponentially. Using an open source solution, companies are no longer forced to pay licensing fees for software or premiums for proprietary hardware and software.

While companies will pay less for licensing, they will continue to receive the Enterprise-grade support they get from traditional vendors, including software additions, service level agreements, and additional services such as training. If there is a failure, companies will have the ability to call and resolve the issue. Guaranteed service contracts will make open source the low risk solution for wide adoption in an Enterprise environment.

Hits: 9115
file icon Overview of H.323 - SIP Interworking (Radvision)hot!
16.05.2001
Looking at IP networks and VoIP, it is clear that both protocols - SIP and H.323 - are widely deployed and here to stay. For carriers wishing to provide global VoIP services, the interconnection of these disparate networks into a unified VoIP network is an urgent and high priority. The SIP-H.323 InterWorking Function (IWF) has been defined as a logical entity that allows the interworking between the SIP and H.323 protocols. This includes call sequence mapping, message parameter mapping, translation between H.245 and SDP, state machines, and the handling of different call procedures.

This paper looks at the issues and challenges of the IWF and presents a number of approaches for achieving the necessary functionality.

Hits: 4425
file icon QoS Testing in the VoIP Environment (Empirix)hot!
14.03.2006

In recent years, the business world has reaped tremendous benefits from the many exciting products and applications made possible by the marriage of data and voice technologies. Now, the efficiencies and enhanced services resulting from that revolution are being enhanced by the magnitude of change possible as data networks become the transport for voice. IP Telephony, or Voice over IP (VoIP), is the exploding new technology enabling voice to be carried over IP-based, packet-switched local and wide area networks.

Now the efficiencies and enhanced services resulting from that revolution are being eclipsed by the magnitude of change possible as data networks become the transport for voice. The open standards approach adopted within the voice over IP world has meant that public and private network operators are now able to remove their dependence on buying everything from one vendor that was a fact of life in the closed/proprietary nature of the traditional circuit- switched world. Other advantages that have attracted a lot of attention include the ability to bypass the PSTN and its toll charges,applications made possible by merging voice with Internet data,lower operating expenses,and network service presence.

Hits: 3372
file icon Robustness testing of SIP implementations (Codenomicon)hot!
18.11.2003
Introduction

SIP[4] will be a part of future IP telephony infrastructure, raising new security concerns and issues. As past experience has shown, a majority of these problems are due to software vulnerabilities. One class of software vulnerabilities is introduced during implementation, in the coding process, due to non-robust handling of input data. An example listing of vulnerability types and failures is in Table 1.

One way to counter these vulnerabilities is to do white box testing and to perform code audits. This white paper discusses this process.

Hits: 3432
file icon sbmain_elab_NAT_for_sip_centerhot!
05.04.2006
NAT traversal with SIP
Homepage: www.sipknowledge.com/elearn.html
Hits: 2027
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Bernard Hammer is responsible at Siemens for technical and strategic standardization issues in the field of IP-based communications for enterprise networks.

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