network design

For more sample High Level Diagrams, please check these links

LAN Diagrams Templates

http://www.conceptdraw.com/diagram/diagram-for-lan

A Basic LAN network Architecture

http://www.excitingip.net/27/a-basic-enterprise-lan-network-architecture-block-diagram-and-components/

Voice Design:

View Slide Show VoIP

http://www.topdownbook.com/

 

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.

http://www.data-compression.com/speech.shtml#demo

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".

http://www.erlang.com/calculator/eipb/

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 likely to be a new chapter on video applications, which I was a little surprised was not covered more in the text.

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 value 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.

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