Implementing Cisco Voice Gateways and Gatekeepers (GWGK) v © Cisco musicmarkup.info Cisco Voice Gateways and Gatekeepers Understanding and configuring GW/GK in complex VoIP networks Denise Donohue, CCIE® No. Editorial Reviews. From the Back Cover. Cisco Voice Gateways and Gatekeepers . Understanding and configuring GW/GK in complex VoIP networks. Denise.
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The PDF files and any printed representation for this material are the property of Cisco . Implementing Cisco Voice Gateways and Gatekeepers (GWGK) v Implementing Cisco Voice Gateways and Gatekeepers (GWGK) v provides network administrators and network engineers with the knowledge and skills. Implementing Cisco Voice Gateways and Gatekeepers (GWGK) v Copyright © musicmarkup.info
Learn more In this course, you will focus on the legacy gateway and router portions of IP Telephony. Troubleshooting will be addressed as a gateway level including common debug techniques and commands. You'll gain an understanding of converged voice and data networks as it relates to gateway design and deployment. This course includes 30 Cisco e-Lab credits. Your e-Lab credits are good for 90 days after your course ends and can be used for additional practice on the course you just completed or to explore technologies from other courses in the Global Knowledge e-Lab portfolio. Learn more. Every pod has internal and external phones, and just like in a real network, the same simulated public switched telephone network PSTN is accessible through all clusters providing failover scenarios for bandwidth and connectivity problems.
SIP and H. For example, when an MGCP gateway detects a telephone that has gone off hook, it does not know to automatically provide a dial tone. The gateway sends an event notification to the call agent, telling the agent that an off-hook condition has been detected. The call agent notifies the gateway to provide a dial tone. Database services include access to billing information, caller name delivery CNAM , toll-free database services, and calling-card services.
VoIP service providers can differentiate their services by providing access to many unique database services. For example, to simplify fax access to mobile users, a provider can build a service that converts fax to e-mail. Another example is providing a call notification service that places outbound calls with prerecorded messages at specific times to notify users of such events as school closures, wake-up calls, or appointments. Proper supervision of these channels requires that appropriate call connect and call disconnect signaling be passed between end devices.
Correct signaling ensures that the channel is allocated to the current voice call and that a channel is properly deallocated when either side terminates the call. Connect and disconnect message are carried by SIP, H. Each codec type defines the method of voice coding and the compression mechanism that is used to convert the voice stream. Each voice channel reserves 64 kbps of bandwidth and uses the G. In VoIP design, codecs might compress voice beyond the kbps voice stream to allow more efficient use of network resources.
The most widely used codec in the WAN environment is G. It also defines end-to-end call signaling. MGCP provides the signaling capability for less-expensive edge devices, such as gateways, that might not have implemented a full voice-signaling protocol such as H. For example, anytime an event, such as off-hook, occurs on a voice port of a gateway, the voice port reports that event to the call agent.
The call agent then signals the voice port to provide a service, such as dial-tone signaling. SIP defines end-to-end call signaling between devices.
SIP is a text-based protocol that borrows many elements of HTTP, using the same transaction request and response model and similar header and response codes. The end stations telephones that use SCCP are called Skinny clients, which consume less processing overhead.
The H. It was developed as a protocol that provides IP networks with traditional telephony functionality. Today, H. The protocols specified by H. This is achieved by exchanging H. The call-signaling channel is opened between two H. RAS is used to perform registration, admission control, bandwidth changes, status, and disengage procedures between endpoints and gatekeepers.
This signaling channel is opened between an end-point and a gatekeeper prior to the establishment of any other channels.
Because audio is the minimum service provided by the H. Voice gateways are an essential part of VoIP networks, handling the many tasks involved in translating between transmission formats and protocols and acting as the interface between an IP telephony network and the PSTN or PBX.
Gatekeepers and IP-to-IP gateways help these networks scale.
Gatekeepers provide call admission control, call routing, address resolution, and bandwidth management between H. Cisco Voice Gateways and Gatekeepers provides detailed solutions to real-world problems encountered when implementing a VoIP network. This practical guide helps you understand Cisco gateways and gatekeepers and configure them properly. Gateway selection, design issues, feature configuration, and security and high-availability issues are all covered in depth.
The abundant examples, screen shots, configuration snips, and case studies make this a truly practical and useful guide for anyone interested in the proper implementation of gateways and gatekeepers in a VoIP network. Emphasis is placed on the accepted best practices and common issues encountered in real-world deployments.
Cisco Voice Gateways and Gatekeepers is divided into four parts. Part I provides an overview of an IP voice network. Part III addresses voice gatekeepers, including detailed deployment and configuration. Silviu Angelescu. Todd Lammle. Network Warrior. Gary A. Omar Santos. Michael Valentine. Windows Internals, Part 2. Alex Ionescu. David Hucaby. Keith Barker. William Manning.
Data Center Virtualization Fundamentals. Gustavo A. Cisco IOS Cookbook.
Kevin Dooley. Douglas Chick. Joe Casad. Hesham Fayed. Tom Hopkins. Eric Rivard. Understanding IPv6. Joseph Davies. Jim Doherty.
Anthony Sequeira. Cisco ASA. Jazib Frahim. Routing and Switching Essentials Companion Guide. Cisco Networking Academy. Cisco Cookbook. Brian Komar. Kevin Wallace. Allan Johnson. Cisco Router Configuration Handbook. Troy McMillan. Windows Server Hyper-V: Zahir Hussain Shah. William Alexander Hannah. CCIE Voice v3. In addition, wireless "hot spots" in locations such as airports, parks, and cafes that allow you to connect to the Internet might enable you to use VoIP services.
Starting with simple media convergence, these advantages evolved to include call-switching intelligence and the total user experience. Originally, ROI calculations centered on toll-bypass and converged-network savings. This approach results in bandwidth being unused when no voice traffic exists. VoIP shares bandwidth across multiple logical connections, which results in a more efficient use of the bandwidth, thereby reducing bandwidth requirements.
A substantial amount of equipment is needed to combine kbps channels into high-speed links for transport across a network.
Packet telephony uses statistical analysis to multiplex voice traffic alongside data traffic. This consolidation results in substantial savings on capital equipment and operations costs. Service providers can easily segment customers. This helps them to provide different applications, custom services, and rates depending on traffic volume needs and other customer-specific factors. Least-cost routing and time-of-day routing are two examples of advanced call routing that can be implemented to determine the best possible route for each call.
It provides a single user interface for messages that have been delivered over a variety of mediums. For example, users can read their e-mail, hear their voice mail, and view fax messages by accessing a single inbox. These processes include centralized call control, geographically dispersed virtual contact centers, and access to resources and self-help tools.
Encryption of sensitive signaling header fields and message bodies protect packets in case of unauthorized packet interception. A pervasive IP network allows organizations to provide contact center agents with consolidated and up-to-date customer records along with related customer communication.
Access to this information allows quick problem solving, which builds strong customer relationships. Some examples of XML-based services on Cisco IP Phones are user stock quotes, inventory checks, direct-dial directory, announcements, and advertisements. Some Cisco IP Phones are equipped with a pixel-based display that can display full graphics instead of just text in the window. The pixel-based display capabilities allow you to use sophisticated graphical presentations for applications on Cisco IP Phones and make them available at any desktop, counter, or location.
Gateways also provide physical access for local analog and digital voice devices, such as telephones, fax machines, key sets, and private branch exchanges PBX. Unlike a gatekeeper, which in a Cisco environment typically runs on a router, a call agent typically runs on a server platform.
Cisco Unified Communications Manager is an example of a call agent. The videoconference station contains a video capture device for video input and a microphone for audio input.
A user can view video streams and hear audio that originates at a remote user station. Other components, such as software voice applications, interactive voice response IVR systems, and soft phones, provide additional services to meet the needs of an enterprise site.
Migration to VoIP requires an awareness of these required elements and a thorough understanding of the protocols and components that provide the same functionality in an IP network. Voice signaling requires the capability to provide supervisory, address, and alerting functionality between nodes. SS7 uses out-of-band signaling, which, in this case, is the exchange of call control information in a separate dedicated channel.
VoIP presents several options for signaling, including H. SIP and H.