Network LAN Design With VoIP And Wireless Services Sec

Network LAN Design with VoIP, and Wireless Services This section will provide a detailed LAN-design of network with VoIP services, Wireless services, protocols, devices, and interconnectivity, with WAN

I am copying below LAN and VoIP and wireless submittal requirements. Network LAN Design with VoIP, and Wireless Services This section will provide a detailed LAN-design of network with VoIP services, Wireless services, protocols, devices, and interconnectivity, with WAN. This section include but not limited to · Equipment List · Hierarchical IP scheme and VLAN · Link IP addresses · High Level Diagram · Voice and Wireless Design Equipment List: Select all networking hardware Suggested Template Device Cisco Model# Quantity Comments Distribution Switches Cat 3850 Series Swiches port model, wireless capacity/PoE Branch office router RV220W Wireless port switch, built in firewall Access Switches Cat ports IDS Network Monitoring System Fire wall Cisco Unified Call Manger 2 Voice services Step 8: Now name each device as per naming convention (Since it is new company, make it for them) Suggested template Device Device Configured Name Placement Connection Comments Cat 3850 Series switch Data center Switch 1 Data center Distribution Switch VLAN1, 2, 8 The switch has Inventory server, Payroll server, and VLAN 1, 2, 8 Core Switch 7600 Series CoreRouter1 Room#1014 DSW1 and DSW2 ISP1 How to determine quantity of equipment: Always start with minimum number which can meet client requirements. For example, if client has 500 users, then select one switch. Now think other factors such as performance of network, redundancy. Hierarchical IP scheme and VLAN Create an IP Scheme and VLANS. I suggest use the table below to create your hierarchical IP addressing scheme. Location Number of IP Addresses Required Future Growth Rounded Power of 2 Number of Host Bits Subnet Address Assigned Floor.20.0.0-172.20.7.254/21 Floor.20.8.0-172.20.9.254/23 Floor .20.16.0-172.20.16.122/25 Floor4 Create VLAN: In creating VLAN, I will suggest use organizational structure model for simplicity. Examples: VPOPRVLAN1, VPOPRVLAN2 High Level Diagram: Drawing a network topology diagram is the most challenging task. To overcome this challenge, we need to use Cisco modular technology in upgrading the network in other words top down design approach. The top design approach starts with Application, Devices and infrastructure. You will also use the same approach in designing WWTC network. Select all the applications for the network. Then select the devices needed to run these applications. Now you are ready creating network topology diagram. Since, in WWTC network we have one floor, so all of our devices, application and infrastructure will reside in one floor. Create subnets. Generally subnet matches organizational structure. Also, in a large network to increase performance or for security reasons, subnets are created. Furthermore to accommodate the need of a department, the subnets can be subnetted further or VLANs are created or both. Every organization have subnets and VLANs. Let us say we need 20 VLANs, which will serve client’s requirements, performance and security of the network. Assign these VLANs to switches. For example; you need 3 switches to host 3 VLANs for VPOPR. The diagram below depict the scenario. Sample Network Diagram For more sample High Level Diagrams, please check these links LAN Diagrams Templates A Basic LAN network Architecture Voice Design: View Slide Show VoIP 1.Click on design resources 2.Click on Topology Puzzle 3.Click on Click here to start 4.Click on Home Icon (Second from bottom) 5.Click on VoIP Speech Compression Technology Improvements in speech compression technology have reduced the bandwidth required for transmission, but have different effects on voice quality. One of the most widely used compression techniques used is called Conjugate-Structured Algebraic Code Excited Linear Prediction (CS-ACELP). It is incorporated into an international standard (ITU-T G.729) which is rapidly becoming the standard for all VoIP transmission. CS-ACELP is an algorithm that allows for compression of linear encoded 64 kbps voice into approximately 8 kbps. The following link lets you compare several compression techniques including the conventional 64 kbps linear telephone speech, which is used a reference standard, and the CS-ACELP (ITU-T G.729) encoded speech. To my ears, CS-ACELP sounds pretty good compared to the current 64 kbps telephone standard. Click on the link below and then click on the various compression types to compare the voice quality of each type. It is recommended that you use headphones for this. Even an inexpensive set will allow you to hear much better than the PC speakers. Examples of Cisco IP Phone Equipment Since this is a course on Cisco based systems, the following figures show some examples of Cisco IP phone equipment. The Cisco 7912 is a low end desktop phone with a minimal number of features. However, even this low end version supports such features as conference calling, caller ID, call waiting, call pickup, hunt groups, call transfers, etc. The 7912 IP phone plugs into a standard RJ-45 connection directly into a LAN switch. The switch is specially designed to provide both power and a network data connection to the 7912 IP phone. This is called PoE for Power over Ethernet. It is also designed to allow a PC LAN connection to be plugged into the phone. Cisco 7912 IP Phone The Cisco 7920 Mobile IP Phone The 7920 is an IEEE 802.11 wireless phone, which allows free movement within the range of a standard wireless Access Point. Cisco SoftPhone The SoftPhone is a software package that permits calls to be made directly from your PC over the Internet. It is a part of a Cisco package called the Cisco Unified Personal Communicator It is similar to the Skype system. Erlangs and VoIP Bandwidth Calculator The following is a link to simplify Erlang calculations. "This calculator can be used to estimate the bandwidth required to transport a known busy hour traffic figure through an IP based network". Also Cisco has only recently changed the terminology for one of the main components of Cisco IP Telephony. What the text refers to as the "Cisco Unified Communications Manager" was, only a year or so ago, called the "Cisco Unified CallManager" as shown in the figure below . So if you do some background research on Cisco IP Telephony, you will see many, many more references using the old terms Cisco Unified CallManager or simply CallManager instead of Cisco Unified Communications Manager. It was not too long ago, that I set up an IP phone system in the UMUC Okinawa office using the "old" CallManager Express (CME) on a 3725 router. Cisco marketing is constantly changing their terminology to better position their products and I believe that the change from CallManager to Unified Communications Manager is a part of that change. The changes in Cisco terminology is annoying when you are trying to find information on a Cisco product, but it is something you will have to get used to and prepare for if you plan to be involved with Cisco equipment in the future. I am sure that the Cisco engineers are even more annoyed than their customers about changes in terminology. I believe the change to the term Unified Communications Manager is part of a marketing move back towards the old AVVID view of integrated services. Below is a figure from a document from the still recent CallManager "era". As shown, above, the "Unified Communications" includes video as well as voice. AVVID encompassed Voice, Video, and Integrated Data. I believe that Cisco is moving back to the old AVVID theme and the change from the old Unified CallManager terminology to the latest Unified Communications Manager terminology is to support the total integrated management theme. My bet is that the next edition of the CCDA text will change the chapter title from " Voice Network Design Considerations " to Integrated Network Design Considerations " and the Unified Communications Manager will include management of video and integrated data as well as voice or IP Telephony. There is also a new management suite that is already available from Cisco, but has not made it into the CCDA curricula or tests yet. It is called the Cisco Unified Operations Manager as described below. "Cisco Unified Operations Manager is part of the Cisco Unified Communications Management Suite. Operations Manager (OM) uses open interfaces and numerous types of diagnostic tests to continuously monitor and evaluate the current status of both the Unified Communications infrastructure and the underlying transport infrastructure of the network. Operations Manager does not deploy any agent software on the devices being monitored and thus is non-disruptive to system operations." WWTC VoIP Network In designing a VoIP network a designer need to know the number of internal telephones calls and number of outside telephones calls. Outside telephone call means call made outside of organization network. This information can be determined by examining a company’s telephone records for the last 6 months. Also, you can determine the busiest hour and for duration of time. To calculate call duration pick up the longest duration period or calculate the average duration of a call. For WWTC project you can use 70% of phone would be busy in the busiest hours. On the basis of number of calls you need to determine the WAN Bandwidth. After that you will select QoS, communication equipment and IP phones. The information I posted in Week-9 folder will guide you to select phones and equipment. You can correctly calculate the bandwidth of single VoIP call by multiplying the sum of the packet overhead and the packetization size by rate, as shown in the formula below: (overhead+ packetization size) x packet rate = bandwidth required. The process of calculating the necessary bandwidth of a VoIP call is best expressed through an example. First, you must calculate the overhead of the VoIP packet. The Layer 2 and layer 3 header information that is necessary for the transmission of a VoIP packet is considered the packet overhead. The packet overhead is typically the sum of the Layer 2 and Layer 3, header sizes, unless some form of header compression, such as compressed Real-Time Transport Protocol [cRTP] is being used. In this scenario, the VoIP packet is transmitted over a Point-to-Point Protocol [PPP] connection with header compression; therefore, the 6 bytes of PPP header information is considered the Layer 2 overhead, and the 4 bytes of compressed IP, User Datagram Protocol [UDP], and Real-Time Transport Protocol [RTP] header information is considered the Layer 3 overhead. Without cRTP header compression, the Layer 3 overhead would have been 40 bytes, which is the sum of the IP, UDP, and header sizes. Thus the total overhead of the IP packet in this example is 8 bytes, as opposed to 46 bytes for an uncompressed packet. The remainder of the packet is referred to as the voice payload. Next you must determine the packetization size, or voice payload size. The packetization size is dependent on two factors; the codec bandwidth and the packetization period. The codec bandwidth is the number of bits per second [bps] generated by a particular voice codec and is typically expressed in Kbps. The packetization period refers to the number of 10-ms samples that the codec can include in each VoIP packet. Because G. 729 includes two voice samples in each VoIP packet, G. 729’s packetization period is 20 ms. The packetization size derived by multiplying the packetization period by the codec bandwidth. In this scenario, the G. 729 codec has been selected to encode all voice traffic between locations. G. 729 yields a packetization size of 20 bytes. This can be derived by multiplying the codec’s bandwidth of 8 kbps, which is 8,000 bps, by its packetization period of 20 ms, which is 0.02 seconds. Multiplying these values yields 160 bits. Finally, dividing this value by 8, which is the number of bits in a byte, yields a packetization size of 20 bytes. The packetization size is also referred to as the payload size. You should add the overhead to the packetization size to determine the total frame size. Bandwidth is generally expressed in bits per second, not in bytes per second [Bps], so if you performed the overhead and packetization calculations in bytes, as shown in this example, you should multiply the total frame size by 8 to determine the total frame size in bits. For example, after you add the 8 bytes of overhead to the 20 bytes of voice payload, you will have a total frame size of 28 bytes. You should multiply this by 8 to obtain a frame size of 224 bits. Finally, you should multiply the total frame size by the packet rate. The packet rate is the inverse of the packetization period. Cisco VoIP devices have a default packet rate of 50 Packets per second [pps] because their default packetization period is 20 ms. Continuing the G. 729 example from above, G. 729 has a packetization period of 20 because it encodes two 10-ms voice samples in each packet. Thus G.729 requires 50 packets to encode 1 second of voice. The total frame size of 224 bits multiplied by a packet rate of 50 pps yields a bandwidth of 11.2 kbps, as shown below: (overhead + Packetization size) x packet rate = bandwidth (8bytes + 20 bytes) x 50 pps = 1.4 kilobytes/sec 28 x 50 pps = 1.4 Kbps 224 bits x 50 pps = 11.2 Kbps As shown in the equation above, a single VoIP call this circuit will consume 11.2 kbps. Therefore, 11 simultaneous VoIP calls will require 11 times as much bandwidth, which is 123.2 Kbps. Of the choices available, 128 Kbps is the minimum amount of bandwidth that can be used to support the configuration. You can correctly calculate the bandwidth of single VoIP call by multiplying the sum of the packet overhead and the packetization size by rate, as shown in the formula below: (overhead+ packetization size) x packet rate = bandwidth required. In your WWTC Project) voice network, you are required to demonstrate · Bandwidth needed for voice network · Create a high level diagram · Show link bandwidth in high level diagram · Provision for transfer to PSTN network automatically, in case WAN links fail (You need to estimate how many PSTN connections you need for 100% connectivity. Remember, this is your private network and like private IP Addresses, private telephone numbers are not routable on public telephone infrastructure. Wireless Network Design: There is enough information Posted in week-2. VPOPRVLAN1 VPOPRVLAN1 VPOPRVLAN1 V P O P R V L A N 2 V P O P R V L A N 2 V P O P R V L A N 2 VPOPRVLAN3 VPOPRVLAN3 ASW1 ASW2 ASW3 DSW1 DSW.16.0.0/30.1 . .

Paper For Above instruction

The comprehensive design of a Local Area Network (LAN) integrating Voice over IP (VoIP), wireless services, and interconnectivity with Wide Area Network (WAN) components is essential for modern organizational communication infrastructure. This paper presents a detailed overview of such a network architecture, focusing on equipment selection, hierarchical IP scheme, VLAN configuration, high-level topology, and specialized considerations for voice and wireless deployments. The goal is to develop a scalable, secure, and high-performance network that supports current needs and future growth, while ensuring redundancy and reliable connectivity.

Equipment Selection and Network Hardware

Effective LAN design begins with proper hardware selection. For distribution layers, Cisco Catalyst 3850 Series switches are recommended due to their high port density, support for Power over Ethernet (PoE), and advanced management features. These switches will serve as core distribution points, connecting access switches and aggregation points. For branch offices or remote sites, Cisco RV220W routers with wireless capabilities and built-in firewalls are suitable, providing both routing and security. Access layer switches, also from Cisco’s Catalyst series, should encompass ports sufficient to connect endpoints such as IP phones, wireless access points, and PCs. An IDS (Intrusion Detection System), network monitoring tools, and a dedicated firewall should be integrated to safeguard the network perimeter. For voice services, Cisco Unified Communications Manager (CUCM) (formerly CallManager) is the core platform responsible for call control, provisioning, and management. The recommended equipment list includes Cisco IP phones—such as the Cisco 7912 or 7920 wireless IP phones—and softphones for mobility and flexibility. The suggested configuration should support the expected user count while maintaining room for expansion.

Hierarchical IP Scheme and VLAN Design

A hierarchical IP addressing plan facilitates efficient routing and management. A common approach involves subdividing the network into multiple subnets based on physical location and organizational department, with each subnet assigned a Class B /21, /23, or /25 network segment to accommodate growth. For example, a /21 subnet such as 172.20.0.0/21 provides 2048 addresses, suitable for a larger floor or department. Smaller subnets, such as 172.20.8.0/23 or 172.20.16.0/25, can serve specific departments or floors.

VLANs are created in accordance with organizational structure, typically using a naming convention like VPOPRVLAN1, VPOPRVLAN2, up to twenty VLANs as needed. Each VLAN corresponds to a subnet, with assigned IP address ranges and associated ports on switches designed to host specific user groups or device types, such as VoIP phones or wireless access points. This segmentation improves network security, reduces broadcast domains, and enhances traffic management.

Example:

- VPOPRVLAN1: Department A, IP range 172.20.0.0/21

- VPOPRVLAN2: Department B, IP range 172.20.8.0/23

- VPOPRVLAN3: Department C, IP range 172.20.16.0/25

This structure ensures logical traffic separation and simplifies management.

High-Level Network Topology and Diagram

Developing an effective network topology involves a layered, top-down design approach. At the core, a high-capacity Cisco 7600 Series router connects to the ISP via redundant links for load balancing and failover. The core connects to distribution switches (Cisco Catalyst 3850 series), which in turn connect to access switches serving different VLANs. For a single-floor layout, all components reside within that floor, with appropriate physical interconnections mapped visually in a topology diagram.

Security and performance enhancements include deploying multiple switches (ASW1, ASW2, ASW3) for load distribution and redundancy. Wireless access points are integrated into VLANs like VPOPRVLAN1, ensuring wireless devices operate within secured segments.