Converged Network Of Voice And Data example essay topic

Designing A Voice Over IP Telephony Solution November 13, 2003 Abstract When, on March 10, 1876, Alexander Graham Bell uttered the words into a speaker "Watson, come here; I want you". (Encarta October 2003) to his assistant Thomas Watson, he gave birth to the modern communications revolution, since that time the telephone has become an integral part of modern society. With the advent of the modern computer and computer networks humankind was availed the ability to communicate with both words and data. It is a natural progression of technology that these two modes of communication be joined into one single network thus affording a blending of technologies that will enhance our communication abilities. The proposed project detailed in this paper has been prepared with the intention of building a fully converged single network to support; data, voice, and video across all of a medium, sized businesses branch locations. To accomplish this I have researched a Cisco AVVID solution to be deployed at the main facility of a midsize d company.

This document is intended for those technical staff members who will be installing and supporting this network. The headquarters convergence project will build the base infrastructure and provide the initial deployment of an enterprise wide IP telephony system. This work entails the main location only unless specifically referencing other sites for various reasons Table of Contents Abstract 3 Introduction 4 Project Background 4 Project Objective 4 Project Overview and Scope 4 Physical Layer Design 5 Design Requirements and Best Practices 5 Local Area Network (LAN) Quality of Service (QoS) 6 Wide Area Network (WAN) Quality of Service (QoS) 6 Configuration of the Voice Call Processing Server 7 Dial Plan 8 Configuration of the Voice Gateway Hardware and Protocols 9 Transcoding, Conferencing, and MTP Resources 10 Site Survey 10 Pre-installation Site Survey 10 Definition of Completeness 10 Staging Services 11 On-site Installation Requirements 11 Training 13 Documentation 13 Definition of Completeness 13 Project Management 14 Overview 14 Project Control 14 Project Timeline 14 Conclusion 15 Appendix A: Proposal for a Voice Over IP Telephony Solution 16 Appendix B: Progress Report 18 References 20 Final Paper: Designing A Voice Over IP Telephony Solution Introduction When, on March 10, 1876, Alexander Graham Bell uttered the words into a speaker "Watson, come here; I want you". (Encarta October 2003) to his assistant Thomas Watson, he gave birth to the modern communications revolution. Since that time the telephone has become an integral part of modern society. It is a natural progression of technology that these to modes of communication be joined into one single network thus affording a blending of technologies that will enhance our communication abilities.

Project Background The proposed project detailed in this document has been prepared as a precursor to, and first step in, a process that will ultimately result in the installation of a voice over IP solution in a midsize d company. To prepare for the convergence of voice, video and data it is necessary to invest in technology to build an Architecture for Voice, Video, and Integrated Data or AVVID. The main focus of this paper will be the specification and design of that architecture. Network convergence, in any form, needs to deliver reliability, superior performance, be technically advanced; yet easy to use, cost effective and create an infrastructure that allows a network to be prepared for the quick and seamless integration of future applications and solutions. The solution described in this paper, once completed, will provide the necessary foundation for a converged network by successfully installing the necessary data networking components, call processing equipment, and associated IP endpoints.

Project Objective The solution proposed is for a midsize d company and utilizes key data infrastructure pieces such as redundant Catalyst 3550-12 G fiber aggregated switches and Catalyst 3524 PWR access layer switches to create foundational components in the AVVID architecture. The voice solution consists of the Cisco Call Manager call processing application which makes call routing decisions and processes telephony features in the new converged network. The voice main system is the Cisco Unity Unified Messaging system that creates a single point of network access, weather that access is a telephone or a pc, to retrieve both voice mail and email messages and, with an optional fax server, fax messages as well. Project overview and scope It will be necessary to acquire and configure certain hardware and software for this project.

The hardware specified in this document is based on research into the Cisco AVVID best practices. This research was then compared to equipment that is currently installed at the mid sized company that this project is proposed for, to come up with the list below. Physical Layer Design The physical layer design consists of the following equipment: I. IT main server room: a. Two MCS 7835 H Dual hard drive servers for Call Managerb. Two Cisco Call Manager 3.3 software server licenses c. One MCS 7825 H server for Unity Unified Messaging.

One Cisco Unity 4.0 server software licensee for 120 us erse. One Catalyst 3550-12 T 10/100/1000 switch. One Catalyst 3970-24 T 10/100/1000 switch. One existing Cisco 7204 Router with one new 2 port Voice T-1 Port Adapter. One existing Cisco 2620 Router to be deployed as the DSP Farm i. One new IOS IP Plus software image for the 2620 j.

One new NM-HDV-Farm DSP chipset II. Main Campus other locations: a. Thirteen Cisco 7960 XML Display IP Phones. One hundred twenty Cisco 7940 XML Display IP Phones. Four Cisco 7914 side cars to be deployed on two double foot stands with associated local power units.

These are for receptionists. d. Three existing Aironet 1200 wireless Access Points (802.11 B) e. Twenty one Cisco 7920 802.11 B wireless phone sf. One Cisco 7935 IP conference phone. Two VG 248 48 port analog voice gateways h. Two Cisco 3550-24 PWR switches i.

Forty two GBIC Multi-mode short wavelength fiber terminators j. One Cisco 2950 G-12 10-100 with a two GBIC slot switch. One existing Cisco 3745 router with one new NM-HDV-T 1 module for PSTN T-1 circuit and one new S RST 144 phone survivability software license Design Requirements and Best Practices IP telephony refers to the technology for transmitting voice communications over a data network using the standards-based Internet Protocol (IP). According to Alexander, Pearce, Smith, and Whetten "The Cisco Architecture for Voice, Video and Integrated Data (AVVID) provides the infrastructure and features set for creating a single converged network that can handle Voice, Video and data traffic simultaneously. Cisco AVVID provides this capability while maintaining a high level of availability, quality of service (QoS), and security for the network. Cisco AVVID IP Telephony solutions are designed to optimize feature functionality, reduce configuration and maintenance requirements, and provide interoperability with a wide variety of available applications". (p. 105) This section of the paper will introduce the best practices related to a successful IP telephony deployment.

Local Area Network (LAN) Quality of Service (QoS) One of the primary design necessities in a converged network will be the design of a QoS strategy. This QoS strategy will be necessary to guarantee voice quality within the campus infrastructure. This will be accomplished by designing separate queues for all voice traffic. This will virtually eliminate the possibility of dropped voice packets when, for example, an interface buffer fills instantaneously. It will further be necessary to classify or mark traffic as close to the edge of the network as possible. Traffic classification is an entrance criterion for access in the various queuing schemes used within the campus switches.

The design begins with the verification of proper ingress / egress port priority queues configured correctly for the 802.1 p voice priority values. The 802.1 p specification consists of the following: o Develop an endpoint IP addressing and VLAN implementation schema. o Ensure correct spanning-tree implementation for proper rapid fail over of voice paths thru a redundant infrastructure. o Implement a packet coloring policy with Differential Service Code Point (DSCP) values to ensure proper voice packet classification as it traverses layer 2 to layer 3 devices throughout the campus data network. Wide Area Network (WAN) Quality of Service (QoS) To specify an entire list of design best practices for a WAN is beyond the scope of this paper. This is rather a sampling of design practices as they relate to the project described in this paper.

At line speeds above 768 kips, voice priority queuing is required to reduce jitter and possible packet loss if a burst of traffic oversubscribes a buffer. This queuing is similar to the one for the LAN infrastructure. However, at lower speeds (less than 768 kips), it will be necessary to test and implement a technique to minimize the effects of serialization delays. Before placing voice and video traffic on a network the Cisco article Quality of Service Solutions Reference Network Design indicates it will be necessary to ensure that there is adequate bandwidth for all required applications. Once sufficient bandwidth has been provided, it will be necessary to design the WAN to reduce delay, packet loss, and jitter for the voice traffic. WAN QoS features and tools are listed in Table 1 QoS.

(Quality of Service Solutions Reference Network Design) Table 1 QoS: QoS Features and Tools Required to Support IP Telephony for each WAN Technology and Link Speed. WAN Technology Link Speed - 56 kips to 768 kips Link Speed - Greater than 768 kbps Leased Lines o Multilink Point-to-Point Protocol (MLP) o Link Fragmentation and Interleaving (LFIO) o Low Latency Queuing (LLQ) o Compressed Real-time Transport Protocol (c RTP) o LLQFrame Relay (FR) o Traffic Shapingo Llq LFIo c RTP o Traffic Shapingo LLQAsynchronous Transfer Mode (ATM) o TX-ring buffer change so MLP over ATMo LFIo LLQ o TX-ring buffer change so LLQFrame Relay and ATM Service Inter-Working (SIX) o TX-ring buffer change so MLP over ATMo LFIo LLQ o TX-ring buffer change so LLQ Configuration of the Voice Call Processing Server According to Peters and Davidson, "Cisco Architecture for Voice, Video, and Integrated Data (AVVID) for IP telephony is based on a distributed network model for high availability. Cisco Call Manager clusters provide for Cisco Call Manager redundancy. The gateways must be configured with the ability to "re-home" to a secondary Cisco Call Manager in the event that a primary Cisco Call Manager fails. This differs from remote site call survivability in the event of a Cisco Call Manager or network failure". (p. 234) For the headquarters project dual MCS 7835 H servers will be used for redundancy. As other locations are added a cluster will be brought online.

Beginning with the Cisco Call Manager release 3.1, it is possible to cluster as many as 12 servers, of which a maximum of six may run the Cisco Call Manager service that provides call processing. The other servers may be configured as a dedicated database publisher, a dedicated Trivial File Transfer Protocol (TFTP) server, music on hold (MoH) servers, or Computer Telephony Interface (CTI) managers. Media streaming applications such as conference bridge or media termination point may also be installed on a separate server that registers with the cluster. The media streaming servers are in addition to the maximum 12 servers allowed in a cluster. However, there are a number of considerations when considering cluster deployment. o A cluster may contain a mix of server platforms, but the cluster may not contain a mixture of Cisco Call Manager software versions. o All services that are not required on a server should be disabled to maximize the server's capabilities. For example disabling Microsoft IIS on all servers except the database publisher. o All members of the cluster are normally within the same LAN or metropolitan area network (MAN). o Cisco Call Manager Release 3.2 can support up to 1000 H. 323 calls per H. 323 device.

Call admission control (CAC) is an important mechanism to protect a network with QoS enabled from being oversubscribed and unable to provide the level of service required for voice transmission. A specific amount of priority bandwidth for voice must be provisioned on each given link in the path. Call admission control provides mechanisms to control the number of calls between two endpoints. This capability is key to maintaining voice quality for all existing calls and any new ones. Because the network is provisioned to carry a specific amount of voice traffic exceeding this bandwidth will subject the voice traffic to delay, jitter and possibly packet loss. This will result in unacceptable voice quality for all calls.

Dial PlanA well designed dial plan is a vital component of any IP telephony network and all other network elements rely on it in some fashion. According to the Cisco Knowledge Base article, Designing a Dial Plan, a dial plan is essentially IP routing for voice calls as illustrated in figure 1 Dial Plans. IP routing and IP telephony dial plans perform similar functions in that both provide endpoint addressing, alternate path routing, and enforcement of policy restrictions. Figure 1 Dial Plans: One of the fundamental attributes of a dial plan is its ability to route a call transparently to the dialed destination of the physical voice path available. Configuration of the Voice Gateway Hardware and Protocols The Media Gateway Control Protocol (MGCP) provides full support for the hold, transfer, and conferencing.

Because MGCP is a master / slave protocol with Cisco Call Manager controlling all session intelligence, Cisco Call Manager can easily manipulate MGCP gateway voice connections. If an IP telephony endpoint needs to modify the session, for example transfer a call, the endpoint would notify Cisco Call Manager using Simple Gateway Control Protocol (SCCP). Cisco Call Manager then informs the MGCP gateway, using the MGCP User Datagram Protocol (UDP) control connection, to terminate the current RTP stream associated with the session ID and to start a new media session with the new endpoint information. Figure 2, Cisco IP Telephony CIPT, MGCP, illustrates the protocols exchanged between the MGCP gateway, endpoints, and the Cisco Call Manager. Figure 2 MGCP: The MGCP gateway supports supplementary services such as call transfer. Within the MGCP protocol is the concept of packages.

The MGCP gateway loads the DTMF package upon start-up. The MGCP gateway sends symbols over the control channel to represent any DTMF tones it receives. Cisco Call Manager then interprets these signals and passes on the DTMF signals, out-of-band, to the signaling endpoint. All support for MGCP with Cisco Call Manager is limited to Release 3.1 and later. DTMF relay is enabled by default and does not need additional configuration. (Cisco AVVID Network Infrastructure Data-only Enterprise Site-to-Site VPN Design Solutions Reference Network Design) Transcoding, Conferencing, and MTP Resources Cisco Call Manager provides access to a variety of media resources.

A media resource is a software-based or hardware-based entity that performs media processing functions on the voice data streams to which it is connected. Media processing functions include mixing multiple streams to create one output stream, passing the stream from one connection to another, or trans coding the data stream from one compress ion type to another. Cisco Call Manager allocates and uses the following types of media resources: (Lewis, p. 356) o Media Termination Point (MTP) resources o Transcoding resources o Unicast conferencing resources o Music on hold (MOH) resources Site Survey The use of a site survey is essential to the success of this project. As such, the following sections will be used as a guideline in identifying all site preparation work that must be completed. While this paper does not contain a site survey form, it will contain activities to be done in conjunction with the actual site survey form. Pre-installation Site Survey Prior to work beginning it is essential that a pre-installation site survey be done.

This will be necessary to ensure all areas are able to handle the required network load. It will also be necessary to review or generate complete Microsoft Visio drawings of all storage closets, server racks, and wiring closets. Wiring will need to be check to ensure at least Category 5 rating. The Microsoft Exchange server specifications will need to be documented to ensure compliance with the Cisco AVVID standards. It will be also necessary to document the current network including switch location, VLAN and QOS strategies, and WAN connections. A Visio drawing of the current IP schema, sub net mask, WINS / DNS, network server operating system environment, and all gateways currently installed.

Definition of Completeness The site survey will be considered complete when the site survey form, network diagrams, floor diagrams and other drawings are completed. Staging Services When in staging, testing is an essential element and is dependant upon network access for stage two testing. Stage one testing will include the standard bench-level testing only. o Upon arrival all servers will be loaded with Windows 2000 server and the appropriate software as per Cisco specifications. The servers will then be stress tested for 24 hours. o Other equipment will be inspected and assembled per design specification so Power up and execute diagnostic self-test so Load and test the configuration and options according to design specification so Label all component so Record configuration, model, and serial numbers On-site Installation Requirements The following design requirements will be implemented by system engineers and technicians. The list that follows is dependant upon all hardware, software and licensing being in place and available. Furthermore, it will also depend upon all problems listed in the site survey being taken care of and updated to meet the standards outlined under design requirements. o Call Routing and Feature set up: System engineers will set up all call routing and system / phone feature so On-site Installation, integration, and testing will be performed by System Engineers and Technicians o Perform administration and end user training will be provided by system administrators after all system documentation is complete Create detailed design document The key to successfully meeting the target deployment objectives is fully planning and coordinating all responsibilities to ensure a smooth and efficient deployment effort.

On-site installation services include all of the required activities to complete the installation of the solution described in this paper. The service verifies that the system and individual components are operating to design specifications and passes all verification tests. The following statements address both general services and installation of all proposed infrastructure and Call Manager hardware / software with a phased approach to a complete installation. I. Verify successful on site power-up diagnostics II. Connect network and client cables. Call Manager (on-site installation) a.

Install Call Manager Software on the MCS servers. Configure Call Manager using the information gathered during the pre-installation site survey b. Configure analog and digital gateways. The digital PSTN gateways will consist of three Digital PRI T 1 circuits to the PSTN for local and long distance calling. The Call Manager dial plan capabilities will route all local and long distance PSTN calls thru the appropriate gateway. IV.

Install and configure Cisco Unity Unified server V. Configure the Unity Server and verify connectivity to Microsoft Exchange 2000 server for Voice Messaging message store a. Set-up and verify connectivity to Call Managerb. Create and program all voice mailboxes specified in the functional specification c. Create and set up Unity Auto Attendant VI. Install and configure all IP Phones for routing and features as discussed in the site survey visits VII. Place Cisco IP phones based on the extension layout plan V. Verify dial tone on all phones a.

Program all switch modules in the Cisco IP Phones with agreed upon VLAN ID's to accommodate PC's and laptops to be plugged into the additional switch port in each Cisco IP Phone (via 802.1 q trunk ports per each 3550-24 switch port that an IP Telephone and PC will be connected to.) IX. Test station-to-station and station-to-PSTN dialing X. Voice Gateway a. Install and configure all voice gateway modules on new and existing routers to connect to the PSTN XI. Install and configure Two VG 248 analog voice gateways to accommodate up to 96 analog extensions XII. Install and configure an NM-HDV-DSP module for network wide DSP resources in existing Cisco 2620 Router X. Networking Infrastructure. Install dual Catalyst 3550-12 G switches in main distribution rack and connect via fiber to each IDF closet that contains the access layer switch (3524-PWR) b.

Install and configure each of (12) Catalyst 3550-24 PWR access layer switches for 802.1 q trunk ports for each 3550-24 switch port. Configure both voice and data VLANS and DSCP tagging value sc. Program all IDF switch modules in the Cisco IP Phones with agreed upon VLAN ID's to accommodate PC's and laptops to be plugged into the additional switch port in each Cisco IP Phone (via 802.1 q trunk ports per each 3550-24 switch port that an IP Telephone and PC will be connected to) d. Configure correct QoS (Low Latency Queuing) on the DS 3 Circuit interfaces for voice priority between the 7204 Router in the IT main server room site and the 3745 Router at the headquarters closet 2 site.

Configure three Aironet 1200 802.11 b Wireless Access points. Install and configure one Catalyst 2970-24 10/100/1000 in main server room. Configure one existing Catalyst 3550-12 T 10/100/1000 in main server room Training The trainer will review features and functionality of the Call Manager system and IP phone user set-up. Additionally, the trainer will review features and functionality of the Unity Unified Messaging system including end user set-up. It is recommended that a trainer have 10 to 20 users per two hour class.

Documentation Complete documentation will be provided with the following completion documentation pertaining to the converged network solution as described in this paper within 2 weeks of completion of the installation. o Project Charter / Project Plano Detailed Visio Network Diagrams Detailed Design Document Definition of Completeness On-site Installation services are considered complete when the equipment to be installed is unpacked, inspected, self-tested, and the network connection is verified; DCE cables are attached; and basic setup options are performed as designed in the functional specification. Connections will be made for dial access and testing completed. Project Management Overview One project manager will be assigned to oversee the converged-networking project. The project manager will be the single point-of-contact and, will discuss any issues needing resolution, manage change control, and will work to verify milestone dates essential to meet required in-service date. This project will include the following process control activities: o Manage and facilitate efforts of the Voice / Data Networking Solution Track all project tasks to timely completion o Identify and manage changes to the project scope.

Negotiate the impact of such changes to schedule, pricing, contract, etc. Update and sign-off on the Statement of Work to include the new or modified activities related to the final implementation of the project plan (change control). Project Control To effectively determine the degree to which the project plan is being met, the Project Manager (PM) will follow a plan to control and manage this project through a communication plan, change control, and variance management. To maintain change control, the PM will utilize a change control strategy to identify any change, document the change request, screen and assess the change impact, obtain approval, implement the change, and then maintain a log of all change requests and their treatment. Any approved changes to the scope of the project will be reflected in the project plan.

Project Timeline To successfully implement project requirements, will require an estimated starting date no sooner than 30 days after acceptance of the project plan and a completion date within 90 to 120 days of the start. This is not a serial process; activities will occur concurrently. Conclusion It is impossible to imagine that Alexander Graham Bell could have imagined the impact his device to aid the deaf would have on the world. There are few inventions that change the world and the way we do business and live our personal lives - the telephone is one of those inventions. The next device that absolutely changed our world was the personal computer.

IP Telephony melds these two technologies, and in doing so brings about the next step in communication. While it will undoubtedly not have the affect that either invention had by itself, it is a step in the evolutionary process. In this paper I have outlined the steps needed to implement an IP telephony solution at a midsize d company and to take a first step in the implementation at my company. If the methodologies presented in this document are followed the company will have a strong LAN and WAN infrastructure and with continued planning and installation, a strong voice over IP Telephony solution. Appendix A: Proposal for a Paper on the Development of A Voice Over IP Telephony SolutionProblemSince the birth of the computer network there have always been two networks that exist in every corporation, a data network and a voice network. The data network is digital and the voice network is analog.

There are very few IT people who have the ability to work with both systems. This necessitates personnel be hired to support both systems. Additionally, each work area must be double wired for telephone and network access. Hard costs are doubled not only in cabling but in other hard costs as well.

Corporations must have a set of switches for data and one for voice. T-1 line charges must be paid for data and separate T-1 lines for voice. Having separate networks can, over time, increase a company's communications charges by as much as 70% over having a converged voice and data network. In addition to hard costs each company must face the soft costs involved in maintaining dual networks. Support personnel costs have already been mentioned but, there are more soft costs that have to be considered. In traditional telephone settings every cubical move and every office move must be reconfigured by a support person.

Traditional telephony makes call centers for a midsize d, regionally diverse company all but impossible. For example, you might have a chain of retail outlets that sells automobile parts. A store in Seattle Washington might be full of walk-in customers and have the phone ringing off of the hook and due to walk-in traffic, employees are unable to answer the call. A branch store in Moses Lake Washington might have no customers and the phone might be idle.

This results in lost customers in Seattle and paid unproductive employees in Moses Lake. Proposed Solution One of the best solutions to this problem is a converged network of voice and data. A converged network will route both voice and date of the same network. This will allow consolidation of personnel, hardware and software onto one redundant network. IP telephony adoption is on the rise industry wide. According to the 2002 Miercom Study by the Allan Sulk an Group, IP telephony will represent 60% of all new voice installations by the year 2006.

IP telephony will enhance productivity, lower total cost of ownership, increase operational agility, and increase scalability. I propose writing a paper on "A Voice Over IP Telephony Solution" that will provide network administrators with an overview of the network architecture and procedures for deploying a voice over IP solution in a mid-sized company. The content of this paper will be primarily concerned with the physical design, including network architecture, hardware, and wiring. This is to be accomplished using site surveys. I then intend to develop a plan for staging and implementing these hardware installations.

I will develop, in some detail, the following areas with regards to hardware parts and design methodologies: o Switches and router so Wiring requirements o Telephone selection Server and software considerations Additionally, I will approach the subject of project management in order to control time and resources. This project is designed to be a first step in the implementation of an IP telephony solution at my company; as such much of the data will be based on my company's needs and their business model. This may be different than another companies needs. Conclusion Voice over IP will provide a more cost effective means of voice and data communication. Currently my company out sources its entire voice network costs. Each location is designed to be a separate entity.

This does not allow for any type of call center activity. Bids to integrate our current system to allow for a call center are $1.8 million. Initial studies in the VOIP solution put those costs at one million dollars. Furthermore, all reoccurring outsourced costs would be brought in house resulting in further savings. This paper on "A Voice Over IP Telephony Solution", will lay the groundwork for the implementation of a Voice and data converged network using Cisco's AVVID technologies. The technical staff reading this paper will have the tools needed to plan for the implementation of a voice over IP solution in a mid-sized company.

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