Showing posts with label Voice Gateway. Show all posts
Showing posts with label Voice Gateway. Show all posts

"The Basics of Voice over Internet Protocol" By Frank M. Groom & Kevin M. Groom


OVERVIEW

Voice over Internet protocol (VoIP), the transmission of person-to-person voice conversations over IP–based networks, is a topic of great interest within the telecommunications industry. With VoIP service, customers see an opportunity to significantly reduce their local and long-distance calling expenses, while long-distance carriers have seen an opportunity to possibly avoid sharing a large portion of their revenues with the local telephone companies for use of their access links to originate long-distance calls. Moreover, service providers and customers see an increasing appeal in the rich array of services that VoIP enables. Among these are a range of Web-like phone displays, information services, and data exchange.

The Basics of Voice over Internet Protocol examines the fundamentals of delivering VoIP service while exploring its potential in the communications market. VoIP is now becoming a mainstream service offered by RBOCs, CLECs, long-distance carriers, international carriers, and virtually any entrant to the transport business. Yet despite this, multiple standards for VoIP still exist, making the interworking of vendor products and the myriad alternative protocols—H.323, SIP, MGCP, etc.—a continuing issue. Further, the quality of VoIP telephone conversations is still under question, and the seeming significant initial costs of setup, including the change-out of phones and PBXs, as well as the addition of gateway and gatekeeper devices, may undermine the benefits to be had from implementing VoIP.

As the industry moves forward, the direction of the complete national network is toward an IP–based architecture, and voice traffic seems to be moving in this common direction. There are many challenges still to be addressed, but the benefits of VoIP continue to draw mounting levels interest, pushing this technology ever further into the mainstream of communications. The Basics of Voice over Internet Protocol analyzes this trend in depth while bringing readers up to date on the evolution of VoIP service.

TABLE OF CONTENTS

Chapter 1: Introduction
The Opportunity for Voice Traffic in IP Networks
Open Standards for VoIP

Chapter 2: Characteristics of Voice and IP Traffic
Public Telephone Network Service versus IP Networking
IP Voice Standards
Basics for Transmitting Voice in IP Packets
VoIP Overhead and Its Effects
The General VoIP Architecture Model
Voice Content Packetization
Voice Packet Transmission
Voice Packet Reception
Standards for VoIP Beyond H.323
Interconnecting VoIP Nets to the Public Telephone Network
Assigning IP Telephone Addresses – DHCP

Chapter 3: VoIP Models for Connection
Enterprise Connection over Public and Private Networks
Residential Connection over Networks
VoIP Terminals

Chapter 4: VoIP Using the H.323 Protocol
The ITU H.323 Protocol
Terminals
Gateways
Gatekeepers
The Operation of an H.323 Network
The H.323 Frame
H.323 Audio Coding and Compression
H.323 Call-Setup Signaling and Message Flow
Operation of Gateways and Gatekeepers in an H.323 Network
Real-Time Transfer Protocol

Chapter 5: SIP for Voice Transmission
SIP Messages
SIP Headers and the TCP/IP Packet
SIP Addressing and Operation
The Specifics of User Agents Using SIP Proxy Servers
The More Detailed Flow Using Proxy and Redirect Servers

Chapter 6: Gateways and Gatekeeper Protocols
Media Gateway Control Protocol – MGCP
The MGCP Commands
Media Gateway Control Protocol – MEGACO
Simple Gateway Control Protocol
Skinny Protocol – An Alternative to MGCP and MEGACO
Summary Comparison of the Protocols

Chapter 7: Transmitting Voice over a Public WAN and IP Network
IP Voice over Frame Relay – IP VoFR
IP Voice over ATM – IP VoATM

Chapter 8: Service-Provider VoIP Offerings
RBOC VoIP Services
CLEC VoIP Services
Long-Distance Carrier VoIP Services
Cable VoIP Services
Small Service-Provider VoIP Services

Chapter 9: Strategies for Vendors, Regulators, and Customers
The Federal Communications Commission
Interexchange Long-Distance Carriers
Regional Bell Operating Companies
Competitive Local-Exchange Carriers
Service Providers
Enterprise Businesses
Residential Customers
VoIP Hardware and Software Vendors

Chapter 10: Conclusion
Appendix: VoIP Terminals and Other Equipment
IP PBX Components and Prices
IP Phones and Prices
H.323 IP Phones and Prices
SIP–Only Phones
General List of IP Phones and Vendors
Cisco Systems Gateway and Module Component Costs
Avaya Gateway and Module Component Costs

FEATURES
· Provides the essential fundamentals of how VoIP functions
· Offers an encompassing overview of the evolution of VoIP services
· Analyzes current market conditions for VoIP and the drivers behind its movement into the mainstream of communications
· Explores the myriad challenges involved in implementing VoIP
· Discusses the various protocols available for VoIP delivery, fully explaining the functionality and benefits of each
· Examines the opportunities and benefits that VoIP presents to RBOCs, CLECs, long-distance carriers, and service providers
· Highlights current VoIP offerings from today's service providers and vendors
· Offers informed strategies on deploying and utilizing VoIP for long-distance carriers, RBOCs, CLECs, service providers, and enterprise businesses, residential customers, and hardware and software vendors
· Provides a comprehensive glossary that defines key terminology
· Presents easy-to-digest graphics, charts, and tables that illustrate important aspects of VoIP

ABOUT THE AUTHORS

Frank M. Groom is a Professor at the Graduate Center for Information and Communication Science at Ball State University. His research is concentrated in the areas of high-bandwidth networking, distributed systems, and the storage of multimedia objects. Dr. Groom is the author of six books-among his best known are "The Future of ATM" and "The ATM Handbook". Dr. Groom received his Ph.D. from the University of Wisconsin - Milwaukee in Information Systems and was formerly Senior Director of Information Systems for Ameritech.

Kevin M. Groom is a Senior Member of the Technical Staff for AT&T, where he is involved in provisioning the network that supports the operations support systems that provide information and functionality to the carrier's long-distance and metro networks. Mr. Groom has a B.A. in Telecommunications and an M.S. in Information and Communication Sciences from Ball State University. He is the co-author of "The Future of IP and Packet Networking" and has presented numerous talks on managing the metropolitan and national carrier networks at networking conferences.

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AudioCodes' “Basic Hybrid” Gateways for Microsoft Office Communications Server 2007

Fortune 500 companies evaluated AudioCodes Media Gateways for the Microsoft Unified Communications Platform Lod, Israel in October 16, 2007. AudioCodes, a leading provider of Voice over IP (VoIP) technologies and Voice Network products, has strengthened its strategic relationship with Microsoft Corp. by introducing a “Basic Hybrid” Media Gateway.

This collaboration has enabled the integration of AudioCodes MediaPackTM and MediantTM Media Gateways with Microsoft unified communications technologies: Microsoft Exchange Server 2007 and Microsoft Office Communications Server 2007.
Fortune 500 companies have evaluated AudioCodes Media Gateways as part of the public beta for Microsoft Office Communications Server 2007 recently released to market. This is in addition to a large number of customers who use the same Media Gateways in conjunction with unified messaging in Microsoft Exchange Server 2007.

“AudioCodes and Microsoft are committed to delivering a unified communications solution that natively operates with existing PBX systems and can directly connect to the PSTN,” stated Lior Aldema, Vice President of Marketing and Product Management at AudioCodes. “This collaboration enables our customers to protect and leverage on existing telephony investments while providing a basis for advancing to the next level of voice communications. AudioCodes Media Gateways enable the enterprise customer to fully utilize the simultaneous strength of both Office Communications Server 2007 and Exchange Server 2007, making use of the same flexible, field -proven, high-quality product.”

“Combining Microsoft’s unified communications software with AudioCodes MediaPack™ and Mediant™ Media Gateways offers customers a rich and integrated communications experience,” said Moz Hussain, director product management, Unified Communications Group, at Microsoft Corp. “Micr osoft’s alliance with AudioCodes provides customers new le vels of business value, helping to create efficient, high quality communication and collaboration experiences.”

AudioCodes offers a range of Media Gateways for the Microsoft unified communications technologies, which includes “Basic” gateways that are designed to interface with external Microsoft Mediation Server and “Basic Hybrid” gateways that integrate the Microsoft Mediation Server software internally.

In order to ease the installation and support of the AudioCodes media gateways in the Microsoft unified communications environment, AudioCodes has created a website landing page http://www.audiocodes.com/Content.aspx?voip=2757, which includes all the technical and marketing resources needed for channel partners and end customers.

The AudioCodes Mediant™ 1000 gateway is specially designed to support these capabilities, in addition to very high VoIP quality, independently tested and acknowledged by ETSI, and interface flexibility and modularity, supporting FXO, FXS, BRI and T1/E1 interfaces in the same box.

AudioCodes gateways are available through an extensive worldwide distribution network that offers local and global toll-free support lines. The distributors may be found by clicking on http://www.audiocodes.com/Content.aspx?voip=2943

About AudioCodes
AudioCodes (NASDAQ: AUDC) provides innovative, reliable and cost-effective Voice over IP (VoIP) technology, Voice Network Products, and Value Added Applications to Service Providers, Enterprises, OEMs, Network Equipment Providers and System Integrators worldwide. AudioCodes provides a diverse range of flexible, comprehensive media gateway, and media processing enabling technologies based on VoIPerfect™ – AudioCodes’ underlying, best-of-breed, core media architecture. The company is a market leader in VoIP equipment, focused on VoIP Media Gateway, Media Server, Session Border Controllers (SBC), Security Gateways and Value Added Application network products.

AudioCodes has deployed tens of millions of media gateway and media server channels globally over the past ten years and is a key player in the emerging best-of-breed, IMS based, VoIP market. The Company is a VoIP technology leader focused on quality and interoperability, with a proven track record in product and network interoperability with industry leaders in the Service Provider and Enterprise space. AudioCodes is a certified member of the Microsoft MSPP program. AudioCodes Voice Network Products feature media gateway and media server platforms for packet-based applications in the converged, wireline, wireless, broadband access, cable, enhanced voice services, video, and Enterprise IP Telephony markets. AudioCodes’ headquarters are located in Israel with R&D in the U. S. Other AudioCodes’ offices are located in Europe, India, the Far East, and Latin America. For more information on AudioCodes, visit http://www.audiocodes.com.

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Interconnecting Voice over IP (VoIP) solutions and legacy fixed or mobile circuit-switched networks: the SMC-SS7 and SMC–3G platforms

Although direct Voice over IP peering is becoming increasingly used by service providers to interconnect their networks, TDM interconnection using the SS7 ISUP protocol is still mandatory in most business models and required by some national regulatory bodies. It is estimated that this type of interconnection will continue to be used for at least the next five years.

This is especially true in the area of Fixed-Mobile Convergence (FMC), where SIP networks must be interconnected to mobile networks. Mobile network operators traditionally use circuit-switched technologies for their telephony services, and will continue to do so even when deploying their 3G infrastructures. So, in the case of video telephony services on mobile 3G phones, for example, SS7 to SIP video gateways are required for the simultaneous reception and emission of voice and video during calls.

The SMC-SS7 is the interconnecting platform between Voice over IP (VoIP) solutions and legacy fixed or mobile circuit-switched networks using SS7 ISUP protocols. SMC-3G adds the extra capacity to interconnect video over IP to circuit switched mobile 3G-324M terminals.

Terminating voice and 3G-324M video telephony calls: a unique feature of the SMC-SS7 platform and its new SMC-3G add-on.

Mobile operators are enabling an increasing number of mobile subscribers to enjoy high-bandwidth Internet and email on their mobile phone through deployment of 3G infrastructures in their network. These services have been particularly successful with business users.

For the consumer mass-market, mobile operators have been using 3G video telephony services to attract subscribers. These services use the 3G-324M protocol, a standard designed for one-to-one video telephony using 64 kbit/s circuit-switched networks. With 3G-324M compliant mobile phones now widely available, mobile video telephony is becoming a reality.

The next step is to increase video telephony usage through the interconnection of 3G-324M circuit-switched mobile networks to SIP packet-based networks, enabling video telephony services between fixed and mobile terminals. The SMC-SS7 platform and its SMC-3G additional software module are pioneering this interconnection by acting as a circuit-to-packet gateway that enables mass market fixed-mobile one-to-one video telephony and interactive video telephony services.

From Netcentrex to Comverse: leading the way in SS7 to SIP interconnect

Netcentrex, a market leader in IP-based communication solutions, and a subsidiary of Comverse since 2006, launched its first SS7 interconnect product in 2001. Constant technical developments have made the SMC-SS7 platform into a comprehensive SS7 telephony interconnect solution with the following features:
  • Configurable ISUP interface
  • SIP and H.323 Voice over IP interface
  • Fully-associated and quasi-associated signaling modes
  • Fully distributed, standards compliant architecture
  • Carrier-Grade platform
  • Easy network integration
  • Supports AudioCodes Media and Signaling Gateways
  • New SMC-3G add-on
  • 3G-324M to SIP interconnect
  • Combined Voice & Video gateway
  • Smooth upgrade path from ISDN to SS7 ISUP
  • Runs on cost-effective hardware platforms
Fully interoperable with AudioCodes’ solutions, the combination of the SMC-SS7 platform with the media and signaling gateways gives operators a head-start in delivery of fixed-mobile communications. They benefit from faster time-to-market, due to pre-integration of the solutions, and from the added value brought by the new communication services in terms of revenue generation and subscriber growth.

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