Whitepapers
As and when any SIP-specific white papers become available they will be added to this page. If you have a SIP-related white paper that you would like to add to this page then please contact us .
It is with the help of our sponsors that we continue to develop The SIP Center resource. You will find here our sponsors whitepapers followed by contributions from many other companies and individuals on a wide range of topics that relate to SIP. We hope there is something helpful here for you!
Interested in contributing, send your request to editor@sipcenter.com!
Is SIP Trunking On Your Horizon?
This white paper from Integrated Research discusses how understanding your call flows, PSTN trunk capacity and usage today will prepare you for a successful SIP trunking implementation in the future.
Multi-site IP Telephony Solutions
The Aculab SIP Bridge - For third party call control
The question of return on investment (ROI) of VoIP
On-board IP architecture - a new approach to computer telephony card design
Best practices for SIP NAT traversal
Managed DNS - Name Management for Next Generation Networks
Media Proxy - NAT traversal for geographically distributed Voice over IP networks
CDR Tool - CDR mediation and electronic invoicing
"Break Free" - Leveraging SIP in Developing Enhanced Applications
Over the last 15+ years, literally thousands of enhanced applications have been developed for the legacy telecommunications infrastructure. From simple voicemail, to sophisticated contact center solutions, these Computer Telephony Integration (CTI) applications have built value on top of basic PSTN dial-tone, generating substantial revenue in both products and services. However, many of the CTI applications were developed using a restrictive and hard-to-learn architecture that limits the developer’s choices in operating systems, choice of technology suppliers and incurs other serious restraints. This whitepaper outlines a migration strategy that leverages SIP to eliminate many of the past restraints and show how to “break free” from the bonds of the legacy CTI architecture.
Building Applications with SIP - the IP Contact Center
The legacy call center has gone through a metamorphosis, emerging as the IP Contact Center which replaces the PBX and separate IVR and ACD systems and merges email and instant messaging into a new architecture that integrates these functions, leveraging Voice over IP technologies. Whether the goal is to reduce costs in the existing call center, leverage inexpensive overseas labor or add Work At Home Agents (WAHA), IP Contact Centers provide tremendous flexibility to adapt to changing markets and labor resources. By reading this whitepaper, you will learn how SIP can be leveraged as a key enabling technology for the IP Contact Center - delivering scalable and cost effective solutions while avoiding restrictive and expensive API development.
Building Applications with SIP - Conferencing / Collaboration Solutions
Global organizations utilize conference calls as a very important business tool for collaboration. Multi-branch organizations were the first to recognize the value in voice and video conferencing services to economize on travel costs and to coordinate business activities. Other smaller organizations have also begun to recognize that having access to easy-to-use conferencing resources speeds collaboration efforts with clients and suppliers. Whether using a tradition TDM PBX, an IP-PBX or a hosted service provider, SIP is seen as a key technology going forward to help tie organizations together and dramatically reduce the costs of conferencing. By reading this whitepaper you will learn how SIP can be leveraged as a key enabling technology for conferencing and collaboration applications - delivering scalable and cost effective solutions while avoiding restrictive and expensive API development.
Telco VoIP Scalability Test Results... for 10 Million Subscribers
SIP Market Overview: An Analysis of SIP Technology and the State of the SIP
VPN Technologies - A Comparison
Managing SIP-based applications with WAN link controllers
Testing ATAs, Gateways, VoIP PBXs, and other Signal Processing Elements in VoIP Networks
To reliably and efficiently handle voice communications, IP networks contain a myriad of signal processing devices including gateways, analog telephone adapters (ATA), and VoIP PBXs. Testing these signal processing elements requires powerful and versatile tools to not only simulate various network conditions but also to accurately measure the resulting output performance. Typical functions the tools must provide are echo, delay, dual tone detection and generation, out of band features, jitter buffer loading, packet concealment algorithms, voice activity detection, various codecs, and applicable protocols such as RTP and RTCP. Signaling protocols such as SIP, H323, MGCP, and Megaco may also be required to test any interactions between signaling and media. In the final analysis, overall voice quality must be assessed with these devices performing their functions. Testing should encompass functional verification, statistical variation such as light to and heavy loading, anomalous conditions such as impairments from TDM and IP sides, and stability testing for reliability and performance.
Sip Trunking Benefits and Best Practices
A SIP trunk is a service offered by an ITSP to use SIP to set up communications between an enterprise PBX and the ITSP. A trunk includes multiple voice sessions – as many as the enterprise needs. 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.
(www.ingate.com) Solving the Firewall/NAT Traversal Issue of SIP: Who Should Control Your Security Infrastructure?
Interaction Center Platform® Architecture: Executive Overview
IP Telephony and the Interaction Center Platform®
Examining the Value of SIP in the Enterprise
SIP: The Global Standard that Enables Businesses to Leverage the Power of the Internet
Naval Research Laboratory
Multi-Service Domain Protecting Interface Architecture
The Multi-Service Domain Protecting Interface (MSDPI) defines a technology that functions as a secure interface between the network (Cypher Text Domain) and the encryption engine. The technology demonstrated in this report will provide many benefits to the government and private sector.
Session Initiation Protocol Network Encryption Device Plain Text Domain Discovery Service
This report provides a method for cryptographic isolated domains to discover other cryptographic isolated domains by using the IETF Session Initiation Protocol (SIP). This method, called the SIP Network Encryption Device Plain Text Domain Discovery Service (SIP-DS), will not
require a new IETF standard or any modification to existing IETF standards, nor are any specifically configured infrastructure or network devices required.
IPsec in VoIP Networks White Paper
Session Border Control in IMS - based Converged Networks
Solving the Firewall and NAT Traversal Issues for Multimedia over IP Services
Oracle SDP - Enablers and Future Proofing your Investment PDF)
IMS SIP and Signaling (Register to download) The RADVISION Perspective - A Technology Overview
Improving Quality in IAD and IP Phone Testing
Benchmarking SIP Server Performance
Henning Schulzrinne, Sankaran Narayanan, Jonathan Lennox, Michael Doyle, Columbia University, Ubiquity.
Monitoring and Troubleshooting VoIP Networks with a Network Analyzer
Promiscuous Monitoring in Ethernet and Wi-Fi Networks
IMS Converged Services Gateway