SIP Trunking

SIP Trunking

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by Christina Hattingh, Darryl Sladden, ATM Zakaria Swapan
     
 

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The first complete guide to planning, evaluating, and implementing high-value SIP trunking solutions

 

Most large enterprises have switched to IP telephony, and service provider backbone networks have largely converted to VoIP transport. But there’s a key missing link: most businesses still connect to their service providers via

Overview

The first complete guide to planning, evaluating, and implementing high-value SIP trunking solutions

 

Most large enterprises have switched to IP telephony, and service provider backbone networks have largely converted to VoIP transport. But there’s a key missing link: most businesses still connect to their service providers via old-fashioned, inflexible TDM trunks. Now, three Cisco® experts show how to use Session Initiation Protocol (SIP) trunking to eliminate legacy interconnects and gain the full benefits of end-to-end VoIP.

 

Written for enterprise decision-makers, network architects, consultants, and service providers, this book demystifies SIP trunking technology and trends and brings unprecedented clarity to the transition from TDM to SIP interconnects. The authors separate the true benefits of SIP trunking from the myths and help you systematically evaluate and compare service provider offerings. You will find detailed cost analyses, including guidance on identifying realistic, achievable savings.

 

SIP Trunking also introduces essential techniques for optimizing network design and security, introduces proven best practices for implementation, and shows how to apply them through a start-to-finish case study.

 

  • Discover the advanced Unified Communications solutions that SIP trunking facilitates
  • Systematically plan and prepare your network for SIP trunking
  • Generate effective RFPs for SIP trunking
  • Ask service providers the right questions—and make sense of their answers
  • Compare SIP deployment models and assess their tradeoffs
  • Address key network design issues, including security, call admission control, and call flows
  • Manage SIP/TDM interworking throughout the transition

 

This IP communications book is part of the Cisco Press® Networking Technology Series. IP communications titles from Cisco Press help networking professionals understand voice and IP telephony technologies, plan and design converged networks, and implement network solutions for increased productivity.

Product Details

ISBN-13:
9781587059476
Publisher:
Pearson Education
Publication date:
02/18/2010
Series:
Networking Technology: IP Communications
Sold by:
Barnes & Noble
Format:
NOOK Book
Pages:
360
File size:
4 MB

Meet the Author

Christina Hattingh is a member of the technical staff in the Access Routing Technology Group (ARTG) of Cisco. The ARTG router product portfolio, including the Cisco 2800, 3800, 2900, and 3900 Series integrated services routers and their predecessors, was one of the first Cisco platforms to converge voice and data starting in the late 1990s by offering TDM voice interfaces, WAN interfaces, and critical QoS features. Over time sophisticated call control and routing elements were integrated into the router-based platform making stand-alone VoIP deployments and wide inter-vendor VoIP network interoperability possible. In this role, Christina trains Cisco sales staff and customers and consults widely on voice network deployment and design. She is a long-time speaker of the Cisco Networkers and CiscoLive conferences. Christina holds a graduate degree in mathematical statistics and computer science from the University of Pretoria, South Africa.

 

Darryl Sladden is a product manager at Cisco and has been with Cisco for more than ten years. Currently, Darryl is a member of the ARTG at Cisco. The ARTG responsibilities include the Cisco ISR and ISR G2, AS5000, and the Cisco Unified Border Element (CUBE). Darryl has been a key contributor to the AS5000 product, CUBE, and several other VoIP technologies at Cisco for several years. The CUBE and the AS5000 product lines are widely used by service providers and enterprise customers as border elements between SIP, H.323, and TDM networks. Darryl has worked with many service provider and enterprise customers who use CUBE to implement SIP Trunks into both Cisco Unified Communications Manager (CUCM) and Cisco Unified Communications Manager Express (CUCME) solutions. Darryl has a degree in mathematics from the University of Waterloo and holds a patent in the use of voice-based network management, and several other patents are under consideration.

 

ATM Zakaria Swapan is a member of the technical staff in the ARTG at Cisco. The ARTG responsibilities include the Cisco 2800, 3800, 2900, and 3900 Series integrated services routers and the CUBE. ATM has been a key contributor to SIP, Secure Unified Communications, Wireless Voice, Network Intelligence, Network Virtualization, RSVP, and many other developments. ATM has also worked with service providers and enterprise customers who use CUBE to implement SIP Trunks into both CUCM and CUCME solutions. ATM holds an M.S. degree in computer science from Texas A&M University and a B.S. degree in computer science and engineering from Bangladesh University of Engineering and Technology (BUET).

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SIP Trunking 3.6 out of 5 based on 0 ratings. 7 reviews.
Boudville More than 1 year ago
What's so great about SIP and why should you [ie. your company] migrate to it? The book explains at various levels the reasons. Succinctly, you should look at Chapter 4, which has an easily understood thread. One main reason is simply to reduce toll charges for long distance, international and local access calls. If you have already informally used VoIP, then going to SIP is essentially a corporate equivalent of moving to it. To some readers, the best reason for SIP is what you can then avoid dealing with your local phone company, which the book characterises as often inflexible and giving poor service. Typically, SIP leads to a more efficient use of your Internet bandwidth. When there are few [incoming or outgoing] calls using SIP, then that "unused" bandwidth is available for general Internet access by your users and by visitors to your website. The book also briefly mentions Power of Ethernet, where a device using this has just one cable that carries both an Internet connection and power. Eliminating the extra power cable can be useful. Unfortunately, the book is marred by a poorly edited first chapter. This chapter was hastily written and not proofread. The ITU is not the Internal [sic] Telecommunications Union. While IEFT in one section is meant to be IETF. Sadly, on page 2, we see nonsense like "200 to 2000 khz", "20 kz to 20,000 hkz" and "20 kz to 20,000 kHz". The chapter will be most readers' initial impression of the book, and this is just sloppy.
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