IT Professionals: January 2009

Friday, January 30, 2009

CCNP Self-Study ONT Official Exam Certification Guide.pdf

Description

As a final exam preparation tool, the CCNP ONT Quick Reference Sheets provide a concise review of all objectives on the new CCNP ONT exam (642-845). This digital Short Cut provides you with detailed, graphical-based information, highlighting only the key topics in cram-style format.

With this document as your guide, you will review topics on voice-over-IP (VoIP) networking, quality of service (QoS), IP QoS mechanisms for implementing the DiffServ QoS model, AutoQoS, wireless security, and basic wireless management.

These fact-filled Quick Reference Sheets allow you to get all-important information at a glance, helping you to focus your study on areas of weakness and to enhance memory retention of essential exam concepts.

With this document as your guide, you will review topics on implementing a voice-over-IP (VoIP) network, implementing quality of service (QoS) on converged networks, and implementing the DiffServ QoS model with specific IP QoS mechanisms, AutoQoS, wireless security, and basic wireless management.

Table of Contents:
Chapter 1: Network Architecture
Chapter 2: Cisco VoIP
Chapter 3: QoS Overview
Chapter 4: QoS Details
Chapter 5: AutoQoS
Chapter 6: Wireless Scalability

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CCNP Self-Study ONT Official Exam Certification Guide.pdf

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Code: Cisco CCVP - Exam-Pack: 642-432 CVOICE

Cisco Voice over IP
Trainer: Jeremy Cioara (Trainer Comments)
Running Time: 11 Hours

Code: This training provides the core knowledge of Voice over IP (VoIP) required for implementing and supporting data and voice integration solutions at the network level. It helps you grasp the VoIP vision, and to answer three major questions most companies have about VoIP solutions:

* Why are so many companies moving to VoIP?

* What are the benefits of using VoIP?

* Is VoIP right for our company?

Once the CVOICE videos help you answer these three questions, they get into the nitty-gritty of VoIP implementation. They teach you how to move an existing PBX or Key System based network to the newer VoIP technology without getting rid of all your old equipment, so you can maximize the cost efficiency of your VoIP implementation. They teach you how to deploy a worldwide VoIP infrastructure, even using lower-end equipment. The videos also cover techniques that allow you to configure your Cisco router to continually monitor the WAN line quality and dynamically change connections as necessary.

"CVOICE [Exam-Pack: 642-432 from CBT Nuggets] not only prepares you for the Cisco exam, but also has multiple practical VoIP deployment videos showing exactly how an administrator would set this up in the real world," explained instructor Jeremy Cioara.

The videos cover all objectives for the Cisco CVOICE exam 642-432, which is required for CCVP, Cisco IP Telephony Express Specialist and Cisco IP Telephony Support Specialist certifications. A thorough understanding of Cisco networking such as CCNA or equivalent knowledge is recommended before viewing these videos. The training contains 19 videos providing more than 11 hours of instruction.

Contents: Cisco CCVP Series - Exam-Pack: 642-432 CVOICE
- IP Telephony: The Vision and Cisco Certification (free video!)

- The World Today: Traditional Telephony

- The World Tomorrow: Packetized Telephony

- How Spoken Voice Becomes a Packet

- Home and Small Office Connections: Understanding Analog Interfaces

- Home and Small Office Connections: Configuring Analog Interfaces

- Medium to Large Office Connections: Understanding Digital Interfaces

- Medium to Large Office Connections: Configuring Digital Interfaces

- Buying the Right Stuff: What Router, What Interfaces?

- Dial-Peers: The Rest of the Story, Part 1

- Dial-Peers: The Rest of the Story, Part 2

- Dial-Peers: The Rest of the Story, Part 3

- Voice Protocols: H.323 Concepts and Configuration

- Voice Protocols: SIP Concepts and Configuration

- Voice Protocols: MGCP Concepts and Configuration

- Quality of Service: The Missing Piece, Part 1

- Quality of Service: The Missing Piece, Part 2

- The Good Busy Signal: Call Admission Control

- Wrap-Up: The Complete Network Design

Code:

Code: pass = sad_gull

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Thursday, January 29, 2009

Cisco - CCVP Cisco Voice over IP CVOICE 3rd(2008).pdf

CCVP Certification

The CCVP certification recognizes the increased importance placed on IT professionals of today who are responsible for integrating voice technology into underlying network architectures. Individuals who earn a CCVP certification can help create a telephony solution that is transparent, scalable, and manageable. Earning a CCVP certification validates a robust set of skills in implementing, operating, configuring, and troubleshooting a converged IP network. The certification content focuses on Cisco Systems CallManager, quality of service (QoS), gateways, gatekeepers, IP phones, voice applications, and utilities on Cisco routers and Cisco Catalyst switches.

Contents at a Glance

Introducing Voice over IP Networks 3
Considering VoIP Design Elements 55
Routing Calls over Analog Voice Ports 125
Performing Call Signaling over Digital Voice Ports
Examining VoIP Gateways and Gateway Control
Identifying Dial Plan Characteristics 321
Configuring Advanced Dial Plans 367
Configuring H.323 Gatekeepers 441
Establishing a Connection with an Internet Telephony Provider 521

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Wednesday, January 28, 2009

Understanding WAN Acceleration Techniques

Recent advances in technology have enabled WAN acceleration to transition from a tactical fix to a strategic IT investment. However, these advances have also led to increased confusion in the marketplace.
Never before have there been so many WAN acceleration products available, leveraging a wider variety of technologies, with varying levels of results.
Partly as a result of this, and the relative newness of the technology, there is a lot of confusion and uncertainty by about some of the more common WAN acceleration technologies being employed today. This article will look at how these technologies work and why they are important.

Data reduction
The most efficient way to accelerate the transfer of information across the WAN is to not send it in the first place. This is the major principle employed by data reduction, a new WAN acceleration technology that provides significant benefits in the form of increased WAN bandwidth efficiency and reduced application response time.
In a data reduction scenario, acceleration appliances examine all data in real-time prior to it being sent across the WAN. This information is stored in local data stores on each appliance.
Whenever duplicate information is detected, references are sent to the appropriate appliance instructing the device to deliver the information locally (instead of re-sending it across the WAN). By preventing repetitive information from traversing the WAN, data reduction can reduce over 90 per cent of WAN bandwidth. By delivering information from local data stores, data reduction helps to provide LAN-like performance across the WAN.

Compression
There are various types of compression used, and the term itself refers to a technique employed to reduce the bandwidth consumed by traffic traversing the WAN by compressing the files traveling across it.
Payload compression uses algorithms to identify relatively short byte sequences that are repeated frequently over time. These sequences are then replaced with shorter segments of code to reduce the size of transmitted data.
Simple algorithms can find repeated bytes within a single packet; more sophisticated algorithms can find duplication across packets and even across flows.
Header compression can provide additional bandwidth gains by reducing packet header information using specialized compression algorithms.
The gains realized by compression techniques vary depending on the mix of traffic traversing the WAN, but are fairly consistent across different vendors' solutions. Text and spreadsheets, for example may yield 2-5x compression ratios.
On the other hand, pre-compressed content, like zip files, cannot be compressed much further. Therefore, additional compression does not help these file types.
Enterprises deploying compression technology will typically see around a 50 per cent improvement in WAN utilization, which is the equivalent of doubling the effective WAN bandwidth.
Some additional benefits may be garnered from solutions that apply compression across various flows of traffic (called crossflow compression) and can employ compression techniques on UDP traffic.
VoIP, for example, can significantly benefit from UDP header compression when used in conjunction with other techniques, such as packet coalescing (see below).

Latency mitigation
The time for information to go from a sender to receiver and back is called the latency of the network. For example, since the speed of light is constant, minimum latency is directly proportional to the distance traveled between the two endpoints of communication.
In other words, the longer the distance, the longer the minimum delay. In real-life, the latency is also impacted by queuing and processing delay in routers and other network elements along the path.
Latency often has a big impact on the performance of applications across the WAN. For TCP bulk data transfers, latency can severely limit throughput. This is primarily because TCP congestion control limits the amount of unacknowledged data in transit.
Once the amount of unacknowledged data reaches the congestion window size, transmission of new data is postponed until older data is acknowledged.
Going up the stack there are several acceleration techniques that are used to overcome the latency issues associated with application delivery across a WAN. These include:

TCP acceleration

The TCP protocol was designed to operate reliably over almost any transmission medium regardless of transmission rate, delay, corruption, duplication, or reordering of segments.
However, the introduction of high speed telecommunications links has resulted in ever-higher transmission speeds, which often exceed the domain for which TCP was originally engineered.

CIFS acceleration
Common internet file system (CIFS) is a protocol developed by Microsoft for remote file access that allows most applications to open and share files across the internet or other IP based networks.
Some specific capabilities of CIFS include file access, record locking, read/write privileges, change notification, server name resolution, request batching, and server authentication

Application-specific acceleration

Some vendors perform application-specific latency optimization techniques to improve the performance of specific types of traffic across the WAN, including SQL, HTTP and Microsoft's Messaging API (MAPI). Furthermore, prefetching of content can be used to overcome latency when delivering some applications across the WAN.

Quality of service (QoS)

In an effort to maximize WAN utilization, most enterprises will oversubscribe their WAN links. When demand exceeds the capacity of a WAN link, and traffic is contending for the same limited resource, less important traffic (such as web browsing) may take bandwidth away from business-critical applications.
To prevent this, some WAN acceleration solutions implement quality of service techniques to classify and prioritize traffic based on applications, users, and other criteria.
QoS involves several functions: 1) classification of packets into traffic classes based on characteristics such as source, destination addresses, and/or applications and 2) queuing and service mechanisms that are used to apply service policies based on these classifications, including bandwidth allocation.

Loss mitigation
Even when the physical-layer of a WAN is error-free, some technologies and provisioning practices still lead to packet-loss at the network layer. In fact, it is not unusual to see network packet loss rates as high as 8 per cent in some networks.
When this type of loss is coupled with high latency and the retransmission and congestion-avoidance behavior inherent to TCP, it is not surprising that application performance suffers across a WAN.
Forward error correction (FEC) is a technology that is well known for its ability to correct bit errors at the physical-layer. It works by adding an additional error recovery packet for every 'N' packets that are sent across the WAN.
This FEC packet contains information that can be used to reconstruct any single packet within the group of N.
If one of these N packets happens to be lost during transfer across the WAN, the FEC packet is used on the far end of the WAN link to reconstitute the lost packet.
This eliminates the need to retransmit the lost packet across the WAN, which dramatically reduces application response time and improves WAN efficiency.

Packet coalescing
When packets are small, packet headers consume substantial bandwidth in comparison to the amount of end-user data transferred. Packet coalescing combines multiple user packets traveling between the same two sites into a single coalesced packet.
Used in conjunction with header compression, this amortizes a single header over multiple packets thus decreasing overhead, and therefore bandwidth requirements. Packet coalescing is particularly beneficial for web applications, VoIP, and interactive applications, like Citrix.

Wide area file services (WAFS)

WAFS is a widely misused term that has sometimes come to mistakenly represent any solution that accelerates the performance of file services. In reality, WAFS is a caching technology that makes it possible to access remote file services as though they were local.
Like other caching solutions, a WAFS appliance simulates an application server, enabling local delivery of specific content. It sits between clients and applications, watching all requests and locally saving copies of the responses.
If another request is made for the same object, the WAFS appliance acts as a proxy, delivering information directly without having to go back across the WAN to the original server.
WAFS enables users to access files services faster, and reduces costs, as remote offices don't need file servers, backup equipment, and additional staff to look after remote office storage concerns. Also remote data is consolidated in the data center and thereby held more securely.
However, WAFS only accelerates file services, and is therefore not as cost effective when compared to other WAN acceleration solutions that also accelerate other key enterprise applications, such as email, web, VoIP, etc.
In addition, because WAFS is using caching technology, it has a potential for coherency issues. In other words, when clients are retrieving and modifying information stored in a local cache, it is easy for this information to get out of synch with information stored in the original application server.

Conclusion
There are many technologies available to address the various issues associated with application delivery across a WAN. The best solution will implement a variety of techniques, both old and new, to improve bandwidth efficiency while reducing perceived application response time.
For more information on WAN acceleration technologies, including detailed examples of how distributed enterprises have deployed these technologies to address application delivery and disaster recovery challenges,

Sunday, January 25, 2009

MCSA/MCSE Self-Paced Training Kit (Exam 70-291): Implementing, Managing, and Maintaining a Microsoft Windows Server 2003 Network Infrastructure

Ace your preparation for the core MCSE networking system exams with in-depth training and practice - all in one box. Now updated for Service Pack 1, these four Microsoft study guides pack the resources you need to help maximize your success on exams 70-290, 70-291, 70-293, and 70-294 - and on the job.Work at your own pace through the lessons, hands-on exercises, troubleshooting labs, and review questions. You get expert exam tips plus a full review section covering all objectives and sub objectives in each study guide. Then use the Readiness Review Suite on CD - featuring advanced technology from Measure Up - to challenge yourself with 1700 questions for self-assessment and practice!

EXAM 70-291

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Friday, January 23, 2009

How VoIP Works

Introduction to How VoIP Works.

If you've never heard of VoIP, get ready to change the way you think about long-distance phone calls. VoIP, or Voice over Internet Protocol, is a method for taking analog audio signals, like the kind you hear when you talk on the phone, and turning them into digital data that can be transmitted over the Internet.
How is this useful? VoIP can turn a standard Internet connection into a way to place free phone calls. The practical upshot of this is that by using some of the free VoIP software that is available to make Internet phone calls, you're bypassing the phone company (and its charges) entirely.
This person is using a computer to talk to a friend in another state.

VoIP is a revolutionary technology that has the potential to completely rework the world's phone systems. VoIP providers like Vonage have already been around for a while and are growing steadily. Major carriers like AT&T are already setting up VoIP calling plans in several markets around the United States, and the FCC is looking seriously at the potential ramifications of VoIP service.
Above all else, VoIP is basically a clever "reinvention of the wheel." In this article, we'll explore the principles behind VoIP, its applications and the potential of this emerging technology, which will more than likely one day replace the traditional phone system entirely.
The interesting thing about VoIP is that there is not just one way to place a call. There are three different "flavors" of VoIP service in common use today:

ATA -- The simplest and most common way is through the use of a device called an ATA (analog telephone adaptor). The ATA allows you to connect a standard phone to your computer or your Internet connection for use with VoIP. The ATA is an analog-to-digital converter. It takes the analog signal from your traditional phone and converts it into digital data for transmission over the Internet. Providers like Vonage and AT&T CallVantage are bundling ATAs free with their service. You simply crack the ATA out of the box, plug the cable from your phone that would normally go in the wall socket into the ATA, and you're ready to make VoIP calls. Some ATAs may ship with additional software that is loaded onto the host computer to configure it; but in any case, it's a very straightforward setup.

IP Phones -- These specialized phones look just like normal phones with a handset, cradle and buttons. But instead of having the standard RJ-11 phone connectors, IP phones have an RJ-45 Ethernet connector. IP phones connect directly to your router and have all the hardware and software necessary right onboard to handle the IP call. Wi-Fi phones allow subscribing callers to make VoIP calls from any Wi-Fi hot spot.

Computer-to-computer -- This is certainly the easiest way to use VoIP. You don't even have to pay for long-distance calls. There are several companies offering free or very low-cost software that you can use for this type of VoIP. All you need is the software, a microphone, speakers, a sound card and an Internet connection, preferably a fast one like you would get through a cable or DSL modem. Except for your normal monthly ISP fee, there is usually no charge for computer-to-computer calls, no matter the distance.

Cisco - 642-825 CCNP ISCW Portable Command Guide(2008).pdf

Format: PDF
Size: 5MB
ISBN: 1587201860

Preparing for the CCNP certification? Working as a network professional?Here are all the CCNP-level commands for the ISCW exam you need in onecondensed, portable resource. The CCNP ISCW Portable Command Guide isfilled with valuable, easy-to-access information and is portable enoughfor use whether you’re in the server room or the equipment closet.This book can help you memorize commands and concepts as you work topass the CCNP ISCW exam (642-825). The guide summarizes all CCNPcertification-level Cisco IOS Software commands, keywords, commandarguments, and associated prompts, providing you with tips and examplesof how to apply the commands to real-world scenarios. Sampleconfigurations throughout the book provide you with a betterunderstanding of how these commands are used in simple network designs.The topics in this portable command guide cover how to do the following:-

Implement basic teleworker services-
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Thursday, January 22, 2009

difference between Layer 3 switch and Router

Layer 3 switch are almost same as a router with following differencesLayer 3 switch is optimized to data as fast as layer2 switch but it makes the decision on how to transmit traffic at layer 3 (just like a router)Perfect in Lan environment as it is built on switching hardware. Another thing is it is faster than a router in LAN environment. Layer 3 use route once switch many policy. It means when it search for specific destination using a routing table, it creates a cache entry in a switch RAM. So next time it does not have to search again for a routing table for same source and destination address. It also picks up hop information from cache. So you can say that Layer 3 switch has the ability to reprogram the hardware dynamically with the current Layer 3 routing information. That is why it is faster than a router.To be frank many of Ciscoâ€™s layer 3 switchâ€™s are actually routers Layer 3 switch good for LANRouters good for WANRouter has some advanced routing functionality which is not available in Layer 3 switch at all.

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Wednesday, January 21, 2009

Google Earth

Google Earth lets you fly anywhere on Earth to view satellite imagery, maps, terrain, 3D buildings and even explore galaxies in the Sky. You can explore rich geographical content, save your toured places and share with others.

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Securing Your VoIP Network

While VoIP being internet-based is a key vulnerability, it also has its beneficial side. The years of experience in fending off or foiling internet attacks is experience that can be used in blocking VoIP assaults; the lessons learned in the data networking field are just as applicable to VoIP networking.
One approach that should be given serious consideration is setting up a separate network for VoIP applications, running in parallel but separate from the data network. This may be considered a serious expense item that is incompatible with the perceived savings from VoIP. On the other hand, one has to consider the potential costs involved if both networks become corrupted or disrupted from an attack on one which also disrupts the other.
Here are some other methods for securing a VoIP network:
– Enable as many of the manufacturer's security protocols as possible, adapting or 'tweaking' these to your own specifications rather than simply following manufacturer's defaults. Keep in mind that hackers and other attackers would probably know these defaults as well.
– Apply a strong authentication and encryption for both data and voice networks. As noted above, use the lessons learned in dealing with data network security problems to establish a preemptive stance in dealing with potential VoIP security concerns.
– Work out access controls and authentication protocols to ensure that only legitimate users can gain access to the VoIP network.
– Use gateway and host-based anti-virus as well as anti-spyware programs to protect crucial VoIP servers. At the same time, consider establishing perimeter security protocols to protect both networks.
A key point to remember is that VoIP is built on already established equipment and applications. The experiences and lessons gained from dealing with security threats to the data network can and should be used in developing security for the voice network.
Conclusion

VoIP's Defensive Linemen

The two primary methods of security for VoIP users are tunneling and encryption. These security measures assist in providing a mechanism of trust in the safe use of the VoIP user's personal data. Most VoIP providers use Layer 2 tunneling and an encryption method called Secure Sockets Layer or SSL to keep hackers at bay. Large corporate enterprises are using similar security mechanisms based on encryption for all internal traffic flowing over the VoIP system as well. It is advisable to route all inbound VoIP traffic that flows via a firewall through a proxy server, thus eliminating any direct connection with the internet.
On a larger level, organizations that are using VoIP as a popular mode of communication rely on a multiple level defense that addresses most VoIP security issues. In this scenario, the VoIP network is divided into secure zones protected by layers of firewall, intrusion prevention, and various additional security mechanisms. The advantage with this strategy is that it allows an organization to logically split and secure separate voice and data networks in front of individual voice and data components and between interactive points within the network. A system (like the one just described) should be complete with authentication, controls access (passwords and firewalls), encryption, an audit trail of calls, and facilities. Recording these issues can prevent security issue to a large degree because they are traceable.

What is VoIP Security?

Any technology that involves transfer of data or information is prone to compromised security. It happens with telephones, cell phones, email and Internet transactions. Because VoIP (Voice Over Internet Protocol) has the internet as its mode of transference it's possible to have your Internet-based called intercepted. To make matters worse, there are techno-troublemakers who are armed with the hacking skills needed to eavesdrop on virtually any call over the Internet they want to. It is impossible to ensure total security on information flow over the web including Internet based phone calls. As new technologies emerge with more highly developed security protocols, there will be those who consider it a unique challenge to crack these online defenses rendering security advances antiquated. The Internet has been notorious for alternating security breaches and accompanying fixes.

As VoIP becomes more popular, VoIP security continues to be stressed as a key to advancement of this technology, especially since it will thrive in the realm of the World Wide Web. There are, however, advances in VoIP security that have been utilized by VoIP providers in order to ensure protection of customer's personal information.

What is VoIP?

VoIP (Voice over Internet Protocol) is simply the transmission of voice traffic over IP-based networks.
The Internet Protocol (IP) was originally designed for data networking. The success of IP in becoming a world standard for data networking has led to its adaption to voice networking.

The Economics of VoIP
VoIP has become popular largely because of the cost advantages to consumers over traditional telepone networks. Most Americans pay a flat monthly fee for local telephone calls and a per-minute charge for long-distance calls.
VoIP calls can be placed across the Internet. Most Internet connections are charged using a flat monthly fee structure.
Using the Internet connection for both data traffic and voice calls can allow consumers to get rid of one monthly payment. In addition, VoIP plans do not charge a per-minute fee for long distance.
For International calling, the monetary savings to the consumer from switching to VoIP technology can be enormous.

VoIP Telephones

There are three methods of connecting to a VoIP network:
Using a VoIP telephone
Using a "normal" telephone with a VoIP adapter
Using a computer with speakers and a microphone

Types of VoIP Calls
VoIP telephone calls can be placed either to other VoIP devices, or to normal telephones on the PSTN (Public Switched Telephone Network).
Calls from a VoIP device to a PSTN device are commonly called "PC-to-Phone" calls, even though the VoIP device may not be a PC.
Calls from a VoIP device to another VoIP device are commonly called "PC-to-PC" calls, even though neither device may be a PC.

What is the difference between a GRE tunnel and IPsec tunnel?

Generic Routing Encapsulation (GRE), defined by RFC 2784, is a simple IP packet encapsulation protocol. GRE is used when IP packets need to be sent from one network to another, without being parsed or treated like IP packets by any intervening routers.
For example, in Mobile IP, a mobile node registers with a Home Agent. When the mobile node roams to a new network, it registers with a Foreign Agent there. Whenever IP packets addressed to the mobile node are received by the Home Agent, they can be relayed over a GRE tunnel to the Foreign Agent for delivery. It does not matter how the Home Agent and Foreign Agent communicate with each other -- hops in between just pass along the GRE packet. Only the GRE tunnel endpoints -- the two Agents -- actually route the encapsulated IP packet.
The IP Security (IPsec) Encapsulating Security Payload (ESP), defined by RFC 2406, also encapsulates IP packets. However, it does so for a different reason: to secure the encapsulated payload using encryption. IPsec ESP is used when IP packets need to be exchanged between two systems while being protected against eavesdropping or modification along the way.
For example, in a site-to-site VPN, a source host in network "A" transmits an IP packet. When that packet reaches the edge of network "A" it hits a VPN gateway. VPN gateway "A" encrypts the private IP packet and relays it over an ESP tunnel to a peer VPN gateway at the edge of network "B." VPN gateway "B" then decrypts the packet and delivers it to the destination host. Like GRE, it doesn't really matter how the two VPN gateways communicate with each other -- hops in between just pass along the ESP packet. But unlike GRE, someone at those hops could not possibly look at or change the encapsulated IP packet, even if they wanted to. That's because cryptographic algorithms have been applied to scramble the IP packet and detect any modification or replay.
In summary, use GRE where IP tunneling without privacy is required -- it's simpler and thus faster. But use IPsec ESP where IP tunneling and data privacy are required -- it provides security features that are not even attempted by GRE.

OSI Reference Model illustrated.

OSI (Open Systems Interconnection) is a standard description or "reference model" for how messages should be transmitted between any two points in a telecommunication networks. Its purpose is to guide product implementors so that their products will consistently work with other products. The reference model defines seven layers of functions that take place at each end of a communication. Although OSI is not always strictly adhered to in terms of keeping related functions together in a well-defined layer, many if not most products involved in telecommunication make an attempt to describe themselves in relation to the OSI model. It is also valuable as a single reference view of communication that furnishes everyone a common ground for education and discussion.
Developed by representatives of major computer and telecommunication companies beginning in 1983, OSI was originally intended to be a detailed specification of interfaces. Instead, the committee decided to establish a common reference model for which others could develop detailed interfaces, that in turn could become standards. OSI was officially adopted as an international standard by the International Organization of Standards (ISO). Currently, it is Recommendation X.200 of the ITU-TS.
The main idea in OSI is that the process of communication between two end points in a telecommunication network can be divided into layers, with each layer adding its own set of special, related functions. Each communicating user or program is at a computer equipped with these seven layers of function. So, in a given message between users, there will be a flow of data through each layer at one end down through the layers in that computer and, at the other end, when the message arrives, another flow of data up through the layers in the receiving computer and ultimately to the end user or program. The actual programming and hardware that furnishes these seven layers of function is usually a combination of the computer operating system, applications (such as your Web browser), TCP/IP or alternative transport and network protocols, and the software and hardware that enable you to put a signal on one of the lines attached to your computer.
OSI divides telecommunication into seven layers. The layers are in two groups. The upper four layers are used whenever a message passes from or to a user. The lower three layers (up to the network layer) are used when any message passes through the host computer. Messages intended for this computer pass to the upper layers. Messages destined for some other host are not passed up to the upper layers but are forwarded to another host. The seven layers are:
Layer 7: The application layer...This is the layer at which communication partners are identified, quality of service is identified, user authentication and privacy are considered, and any constraints on data syntax are identified. (This layer is not the application itself, although some applications may perform application layer functions.)
Layer 6: The presentation layer...This is a layer, usually part of an operating system, that converts incoming and outgoing data from one presentation format to another (for example, from a text stream into a popup window with the newly arrived text). Sometimes called the syntax layer.
Layer 5: The session layer...This layer sets up, coordinates, and terminates conversations, exchanges, and dialogs between the applications at each end. It deals with session and connection coordination.
Layer 4: The transport layer...This layer manages the end-to-end control (for example, determining whether all packets have arrived) and error-checking. It ensures complete data transfer.
Layer 3: The network layer...This layer handles the routing of the data (sending it in the right direction to the right destination on outgoing transmissions and receiving incoming transmissions at the packet level). The network layer does routing and forwarding.
Layer 2: The data-link layer...This layer provides synchronization for the physical level and does bit-stuffing for strings of 1's in excess of 5. It furnishes transmission protocol knowledge and management.
Layer 1: The physical layer...This layer conveys the bit stream through the network at the electrical and mechanical level. It provides the hardware means of sending and receiving data on a carrier

What is the difference between OSI model and TCP/IP other than the number of layers?

OSI is a standard "reference model" that describes how protocols should interact with one another. Invented by the Department of Defense, TCP/IP became one of the "standards" that enabled the Internet to become what it is today. TCP/IP doesn't map cleanly into the OSI model, but it is convenient to think in terms of the OSI model when describing protocols.
The major differences between the OSI and TCP/IP are:
The application layer in TCP/IP handles the responsibilities of layers 5, 6 and 7 in the OSI model.
The transport layer in TCP/IP does not always guarantee reliable delivery of packets at the transport layer, while the OSI model does. TCP/IP also offers an option called UDP that does not guarantee reliable packet delivery.

Firewalls : Troubleshooting Tips - Debug commands

There are many debug commands that you can run to troubleshoot problems on Juniper firewalls. The most common debugs used are:

Debug flow basic shows the flow of traffic through the firewall, allowing for troubleshooting route selection, policy selection, any address translation and whether the packet is recieved or dropped by the firewall.

Debug ike detail show all of the vpn messages between two devices when setting up the vpn. Useful to see problems between Juniper and 3rd party firewalls or configuration porblems.

Other debugs are ones allowing troubleshooting of routing protocols, av, deep inspection, higher availability, NAT and loads of others. You can find out what others there are by typing 'debug ?' on the cli.

Debug captures are sent to a buffer by default but can be sent the console.

Buffer commands:

get dbuf info - Displays debug buffer size in bytes
get dbuf stream - Displays the contents of the debug buffer
set dbuf size - Allocates system memory for the debug buffer
clear dbuf - Clears the contents of the debug buffer

Here are best practices when debug flow basic and how to read the output.

Debug Flow basic

1) get ffilter - see if an filters have been set already, if they have you use 'unset ffilter' to remove

2) set ffilter - allows you to limit the traffic that you capture using src-ip, src-port, dst-ip, dst-port, ip-proto Recommeded as debug flow basic can be intensive on the firewall especially if it is under heavy load.

filters that are written on one line are and 'AND' statement. eg set ffilter scr-ip 10.1.1.1 dst-ip 2.2.2.2 will match 10.1.1.1 AND 2.2.2.2

filters that are on seperate lines are 'OR' statements. eg set ffilter scr-ip 10.1.1.1, set ffilter dst-ip 2.2.2.2 will match 10.1.1.1 OR 2.2.2.2

3) debug flow basic - turns on flow debuging with a level of basic logging

4) clear db - make sure there is nothing in the debug buffer from previous debugs

5) Begin the test, do a ping or try to access the resource that you are having problems with.

6) undebug all or press Esc key - turns off debug

7) get db str - reads the debug buffer and outputs it to the screen for reading.

8) unset ffilter - remove ffilters when finished

Juniper Firewall Packet Process

Packet Process.jpg (51 kb)

Reading a Debug Flow Basic

FW-> get dbuf stream****** 15625.0: packet received [60]****** - This is the interface and zone that the packet arrives on ipid = 34344(8628), @02cdf8b0 packet passed sanity check. ethernet0/5:10.1.1.5/17152->1.1.70.250/256,1(8/0) - This is the packet IP info ports and protocol no session found - Looks for existing sessions on the firewall flow_first_sanity_check: in , out chose interface ethernet0/5 as incoming nat if. - There is NAT on the interface flow_first_routing: in , out search route to (ethernet0/5, 10.1.1.5->1.1.70.250) in vr trust-vr for vsd-0/flag-0/ifp-null [ Dest] 10.route 1.1.70.250->0.0.0.0, to ethernet0/6 - matches route 10 in the trust-vr routing table routed (x_dst_ip 1.1.70.250) from ethernet0/5 (ethernet0/5 in 0) to ethernet0/6 policy search from zone 104-> zone 103 - Tells you which policy set its going to look at zone to zone derived from routing policy_flow_search policy search nat_crt from zone 104-> zone 103 RPC Mapping Table search returned 0 matched service(s) for (vsys Root, ip 1.1.70.250, port 2396, proto 1) No SW RPC rule match, search HW rule Permitted by policy 2 - Tells you which policy the traffic has hit and what the action is No src xlate choose interface ethernet0/6 as outgoing phy if no loop on ifp ethernet0/6. session application type 0, name None, nas_id 0, timeout 60sec service lookup identified service 0. flow_first_final_check: in , out existing vector list 1-4339cf0. Session (id:8061) created for first pak 1 - session created for future packes in the session flow_first_install_session======> route to 1.1.70.250 arp entry found for 1.1.70.250 - firewall arps for where it is going to route packet to. nsp2 wing prepared, ready cache mac in the session make_nsp_ready_no_resolve() search route to (ethernet0/6, 1.1.70.250->10.1.1.5) in vr trust-vr for vsd-0/flag-3000/ifp-ethernet0/5 [ Dest] 12.route 10.1.1.5->0.0.0.0, to ethernet0/5 route to 10.1.1.5 flow got session. flow session id 8061 post addr xlation: 10.1.1.5->1.1.70.250. - shows if there has been and address translation and how the packet looks when it is sent out.

Things to look for when there are problems:

Packet dropped, no route to host - no route has been round in the virtual router that the interface and zone is bound to that the packet arrived on. You need to look in your routing table to check to see if you have a route for the desitnaiton or source depending how you are routing.

Packet dropped, denied by policy - If just before this you see a message saying searching global policy then it means that no policy has been matched so it has been dropped by default. You need to check your policy configuration.

Policy match - Check the policy matach to make sure that it is matching the right polic and not doing allowed or denied by another rule

Also check to make sure any NAT is happening that needs to.

Tuesday, January 20, 2009

Explanation of the Three-Way Handshake via TCP/IP

This article is intended for audiences who are familiar with Transmission Control Protocol/Internet Protocol (TCP/IP) and discusses the process of the TCP three-way handshake that occurs between a client and server when initiating or terminating a TCP connection.

The Transmission Control Protocol (TCP) level of the TCP/IP transport protocol is connection-oriented. Connection-oriented means that, before any data can be transmitted, a reliable connection must be obtained and acknowledged. TCP level data transmissions, connection establishment, and connection termination maintain specific control parameters that govern the entire process. The control bits are listed as follows:
URG: Urgent Pointer field significantACK: Acknowledgement field significantPSH: Push FunctionRST: Reset the connectionSYN: Synchronize sequence numbersFIN: No more data from sender There are two scenarios where a three-way handshake will take place:
Establishing a connection (an active open)
Terminating a connection (an active close) The following sample information was obtained from a Network Monitor capture. Network Monitor is a protocol analyzer that can be obtained from Microsoft Systems Management Server.

For additional information on TCP/IP, please see the following white paper.

Monday, January 19, 2009

What is Microsoft Office SharePoint Server?

Microsoft Office SharePoint Server 2007 is a new server program that is part of the 2007 Microsoft Office system. Your organization can use Office SharePoint Server 2007 to facilitate collaboration, provide content management features, implement business processes, and supply access to information that is essential to organizational goals and processes.
You can quickly create SharePoint sites that support specific content publishing, content management, records management, or business intelligence needs. You can also conduct effective searches for people, documents, and data, participate in forms-driven business processes, and access and analyze large amounts of business data.
Microsoft Office SharePoint Server 2007 Capabilities
Microsoft Office SharePoint Server 2007 provides a single, integrated location where employees can efficiently collaborate with team members, find organizational resources, search for experts and corporate information, manage content and workflow, and leverage business insight to make better-informed decisions.
•
Collaboration Allow teams to work together effectively, collaborate on and publish documents, maintain task lists, implement workflows, and share information through the use of wikis and blogs.
•
Portals Create a personal MySite portal to share information with others and personalize the user experience and content of an enterprise Web site based on the user’s profile.
•
Enterprise Search Quickly and easily find people, expertise, and content in business applications.
•
Enterprise Content Management Create and manage documents, records, and Web content.
•
Business Process and Forms Create workflows and electronic forms to automate and streamline your business processes.
•
Business Intelligence Allow information workers to easily access critical business information, analyze and view data, and publish reports to make more informed decisions.

Routing quiz: prep for your CCNA!

Are you preparing for the CCNA exam? Maybe you just want to test your knowledge of routing and routed protocols. This quiz will tell you if you are up to speed when it comes to TCP/IP protocols.

1. What routing protocol uses cost as its metric?
a. RIP
b. EIGRP
c. OSPF
d. BGP

2. Which of these is an exterior gateway protocol (EGP)?
a. OSPF
b. BGP
c. RIP
d. EIGRP

3. Which of these routing protocols would be used between Autonomous systems (Inter-AS)?
a. OSPF
b. RIP
c. EIGRP
d. BGP

4. Which routing protocol has a default administrative distance of 110?
a. BGP
b. OSPF
c. RIP
d. EIGRP
e. Static

5. Which routing method is most trusted by a router?
a. BGP
b. IGRP
c. Static
d. RIP
e. OSPF

6. What is the process of sending routes from one routing protocol to another routing protocol?
a. Redistribution
b. Distribution
c. Filtering
d. Prepending
e. Source routing

7. Which command tells RIP to send hellos, out an interface, to find neighbors and to advertise routes?
a. hello
b. network
c. interface
d. advertise
e. neighbor

8. You want to filter inbound routes from an OSPF neighbor. Which command do you use?
a. Distribute-list
b. Route-list
c. Prefix-list
d. Filter-list

9. In BGP, OSPF and EIGRP, an internetwork under common administration is called what?
a. Routing system
b. Autonomous routing
c. Routing domain
d. Autonomous system

10. Which command do you use to prevent OSPF HELLO packets from going out of a configured interface?
a. no network
b. passive
c. no advertise
d. hello-off
e. quiet

11. In OSPF, every router must have communications back to what?
a. Area zero
b. The root
c. OSPF central
d. Area one

12. Which of these routing protocols would be unavailable on a non-Cisco router?
a. OSPF
b. RIP
c. EIGRP
d. BGP

13. What type of OSPF router would connect an area that is not using OSPF?
a. BDR
b. ABR
c. BR
d. ASBR
e. OSBR

14. A router receives two identical routes from different protocols. What route is put into the routing table?
a. The route with the most trusted AD
b. The best route
c. The OSPF route
d. Both routes

15. If RIP receives two similar routes to a network, which route is put into the routing table?
a. The route with the most trusted administrative distance
b. The most specific route
c. The RIP version 2 route
d. Both routes

ANSWERS
1. C (OSPF)
2. B (BGP)
3. D (BGP)
4. B (OSPF)
5. C (Static)
6. A (Redistribution)
7. B (Network)
8. A (Distribute-list)
9. D (Autonomous system)
10. B (Passive)
11. A (Area zero)
12. C (EIGRP)
13. D (ASBR)
14. A (The route with the most trusted AD)
15. B (The most specific route)

Routing versus Routed protocols

Many networking professionals don't really know what a routing protocol is or does. There are those who also couldn't tell you the difference between a routing protocol and a routed protocol. If you are working as a network engineer or preparing to take the CCNA, you need to understand the distinction between routing and routed protocols, as well as how these protocols function on the network.

Routing protocols
Routing protocols distribute routing information throughout all routers on a network. By knowing about all other routers connected to the network, each router can determine the best path to use to deliver your traffic. Examples of routing protocol would be OSPF, RIP, EIGRP or BGP.

Routed protocols
Routed protocols can be sent over a routed network. Today, with IP (think of TCP/IP) being the predominate protocol in use on the Internet and in most networks, we don't often think about routed versus unrouted protocols. That is because IP is a routed protocol. Meaning, IP is designed such that it can be routed over and through different networks. This is because the IP address is broken up into network and node (host) by the subnet mask. An example of a protocol that is not routed (unroutable) would be NetBIOS. It works great on a small local network (in your house) but wouldn't work well over a large network connected by dedicated network circuits or the Internet.

Difference bwteen Switching and Routing

Difference bwteen Switching and Routing

Routing and switching can be two terms that are difficult to differentiate, so here is a simple explanation that may help to clarify things. First of all switching and routing are not the same thing. Switching involves moving packets between devices on the same network. Conversely, routing involves moving packets between different networks. Switches operate at layer 2 of the OSI Model. A switch, also referred to as a multi-port bridge, is able to determine where a packet should be sent by examining the MAC address within the data link header of the packet (the MAC address is the hardware address of a network adapter). A switch maintains a database of MAC addresses and what port they are connected to. Routers, on the other hand, operate at layer 3 of the OSI Model. A router is able to determine where to send a packet using the Network ID within the Network layer header. It then uses the routing table to determine the route to the destination host.

Routing algorithms and switching techniques

Basic concepts
Router architecture
processing node:
standard bus-based computer augmented with a router.

CAPTION: Generic structure of a processing node
router:
a HW coprocessor implementing lower levels of communication protocol. It comprises of a switch, buffers, and routing and arbitration unit (see Figure 1 ).
external channels:
interconnect routers and define the topology of a connected direct interconnection network.
adjacent nodes:
nodes with directly connected routers.
internal channels:
connect the router to the local processing node and implement the physical HW interface between them.
1-port architecture: one injection and one ejection channel (as in Figure 2),
k-port architecture: k injection and k ejection channels,
all-port architecture: # of injection channels = # of external output channels and # of ejection channels = # of external input channels.

Basic concepts

Basic switching techniques

Performance metrics
channel width w:
# of bits that a physical channel can transmit simultaneously between two adjacent routers. Clearly, it is exactly the size of a phit.
channel rate \mu:
the peak rate in bits/second at which bits can be transfered over each individual wire of a physical channel.
channel bandwidth B:
B=\mu in phits/second, even though it can be defined also as B=w× \mu in bits/second.
channel latency tm:
the inter-router latency of one phit, hence tm=1/\mu=1/B in seconds/phit.
bisection bandwidth B(G)
of a network G: B(G)=B× bwe(G)
startup latency ts:
the time required for packet framing/unframing, copying data between memory and router buffers, validation and so on, at both source and destination side. It depends mainly on the design of system SW, on its interface to user programs, and on the physical interface between processing nodes and routers. It is a static value, independent of network traffic or distance between nodes. It depends only on the size of the message.
network latency:
the elapsed time after the head flit of a packet has entered the network at the source router until the tail flit is ejected. The network latency consists of two components: a conflict-free packet transmission time, which strongly depends on the switching technique, and the blocking time, which includes all possible delays encountered during the lifetime of a packet.

for futher detail visit to:

Friday, January 16, 2009

Free Drivers Download

Welcome to ITProfesional.blogspot.com, the Internet’s longest-running Driver Update website and home to more than 100,000 driver related files. Here you'll discover the information you've been searching for on how to speed up your computer, enhance its performance and obtain the latest driver updates available. You’ll find it all here at ITProfesional.blogspot.com!

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