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Conversation Escalation Make Small Talk Sexy

In this ground-breaking program you'll learn the subtleties of conversation to pinpoint the specific problems that are ruining your chances with women. You'll learn how to draw people out to talk about more interesting topics in a more natural way instead dragging it out of them. And the mindset tricks so that you can Always be in the zone with women whenever you're talking to them. What's unique about this course is that its based on examples and application and is filled with hundred of little bite size game changers that you'll be able to see an immediate impact on your conversations tonight. Read more here...

Conversation Escalation Make Small Talk Sexy Summary

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My Conversation Escalation Make Small Talk Sexy Review

Highly Recommended

Recently several visitors of websites have asked me about this manual, which is being advertised quite widely across the Internet. So I decided to buy a copy myself to figure out what all the publicity was about.

This book served its purpose to the maximum level. I am glad that I purchased it. If you are interested in this field, this is a must have.

Typical Authentication Conversation

Now that you have a better idea of how EAP works, and you have seen the role of a back-end RADIUS server in the authentication process, let's look at a typical authentication conversation with an EAP client, a NAS, and the RADIUS server. There are different implementations of EAP in several different network architectures. Each implementation follows the same basic premise for the authentication flow. Figure 26-2 illustrates a basic authentication packet flow for RADIUS EAP involving a client, a NAS, and a RADIUS server.

Conversation Conventions

Over the years, text users have adopted certain conversation conventions and other shortcuts to make text telephone communication more efficient. Some of these conventions were brought about by the half-duplex nature of the Baudot protocol itself, whereas others have been created to save time and speed up the pace of a typed conversation. The two most notable conventions or initialisms are GA (go ahead) and SK (stop keying). GA is used when a user has finished typing to notify the other party that it is his turn to type. Because the Baudot protocol is half duplex, GA is used to prevent two parties from talking simultaneously. During a Baudot text conversation, interrupting someone and typing while he is typing should not be done because this can garble the transmissions of both parties. conventions are not necessary. However, users will still exercise explicit turn-taking to avoid typing over each during a conversation. SK is typically seen at the termination of a text telephone call....

Introducing Voice over IP Networks

VoIP routes voice conversations over IP-based networks, including the Internet. VoIP has made it possible for businesses to realize cost savings by utilizing their existing IP network to carry voice and data, especially where businesses have underutilized network capacity that can carry VoIP at no additional cost. This section introduces VoIP, the required components in VoIP networks, currently available VoIP signaling protocols, VoIP service issues, and media transmission protocols. Security Mechanisms in an IP network allow an administrator to ensure that IP conversations are secure. Encryption of sensitive signaling header fields and message bodies protect packets in case of unauthorized packet interception.

Definition of Integrated Servie es

Resource Reservation Protocol (RSVP), defined by RFC 2205, is o resource reservation setup protocol for use in on InUServ environment. Specifics of operation ore covered shortly, but Uhe general ideo behind RSVP is UhoU Bob wonUs Uo Uolk Uo SUeve, who is some number of network hops owoy, over on IP video conferencing (IPVC) system. For Uhe IPVC conversation Uo be of accepUooie quo liUy, Uhe conversation needo 384 kbps os bandwidth. Obviously, tue IPVC end

Figure 21 The MAC Address Table Is Initially Empty

The filtering process also means that multiple simultaneous conversations can occur between different devices. For example, if device A and device B want to communicate, the switch sends their data between ports 1 and 2 no traffic goes on ports 3 or 4. At the same time, devices C and D can communicate on ports 3 and 4 without interfering with the traffic on ports 1 and 2. Thus, the overall throughput of the network has increased dramatically.

Example 87 show queueing Monitors WFQ Status on a Router Interface

Router show queueing serial0 Input queue 0 75 0 (size max drops) Total output drops 0 Queueing strategy weighted fair Output queue 3 1000 64 0 (size max total threshold drops) Conversations 1 3 256 (active max active threshold) Reserved Conversations 0 1 (allocated max allocated) (depth weight discards tail drops interleaves) 2 4096 0 0 0 Conversation 1023, linktype ip, length 1504 source 10.1.1.1, destination 10.1.5.2, id 0xC4FE, ttl 120, ToS 0 prot 6, source port 1520, destination port 4563 The following list explains the relevant fields from the output of the show queueing serial0. The shaded portion of Example 8-7 highlights the statistics for a single flow or conversation a conversation queue is the same as a subqueue discussed in the example previously.

The Basics FIFO Queuing

First-in, first-out (FIFO) queuing is the default queuing strategy that applies to all interfaces with more than 2 Mbps, or, in other words, E1 size or greater interfaces. With the FIFO Queuing strategy, packets are forwarded through the interface in the order that they are received. For example,Figure 6-1 shows three traffic conversations, or flows. Conversation A consists of Telnet packet that are approximately 64 bytes packets in conversation B are from a network application, axd range from 750 and 1020 bytes and packets from conversation C are HTTP web traffic packetsu which ame approximately 1500 bytes. When these three host stations send packets during pes riods of low netwoEk traffic utilization, all three conversations should be successful if these same three conversations take place during a period of high network utilization, however, packets from conversation C will be interspersed between the much smaller packets fro m t e A and B conversations, which could potentially...

Displays detailed information about the WFQ system of the selected interface

Output queue 2 1000 64 0 (size max total threshold drops) Conversations 2 4 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) (depth weight discards tail drops interleaves) 1 4096 0 0 0 Conversation 124, linktype ip, length 58 0 (depth weight discards tail drops interleaves) 1 4096 0 0 0 Conversation 127, linktype ip, length 585 The show queue command is used to display the contents of packets inside a queue for a particular interface, including flow (conversation) statistics

Example 811 Output of show policy interface to Monitor CBWFQ

UbrI19-2 show policy int Serial0 service-policy output policy1 class-map class1 (match-all) 0 packets, 0 bytes 5 minute offered rate 0 bps, drop rate 0 bps match access-group 101 Output Queue Conversation 265 Bandwidth 100 (kbps) Packets Matched 0 Max Threshold 20 (packets) (discards tail drops) 0 0 class-map class-default (match-any) 0 packets, 0 bytes 5 minute offered rate 0 bps, drop rate 0 bps match any 0 packets, 0 bytes 5 minute rate 0 bps Output Queue Conversation 266 Bandwidth 200 (kbps) Packets Matched 326 random-detect mean queue depth 0 drops class random tail min-th max-th mark-prob 0 0 0 20 40 1 10 1 0 0 22 40 1 10

Connection Oriented and Connectionless Services

A telephone service is a good example of a connection-oriented service. Before you can start a conversation, the call setup and data transfer phases must be completed. After those stages have completed, you can start talking. When you finish the conversation, the call termination phase takes place. Each of these phases in a telephone call are characteristics of connection-oriented services. Connection-oriented services consist of

Digital Speech Interpolation

Similar to statistical multiplexing, digital speech interpolation (DSI) multiplexes bandwidth among a larger number of users than there are circuits. DSI uses voice activity detection and silence suppression to allocate the silent periods in human speech and put them to active use. Remember that 50 percent of a voice conversation is silence.

Effects of Voice Activity Detection on Bandwidth

Statistically, an aggregate of 24 calls or more might contain 35 percent silence. With traditional telephony voice networks, all G.711 voice calls use 64 kbps fixed-bandwidth links regardless of how much of the conversation is speech and how much is silence. In Cisco VoIP networks, all conversations and silences are packetized. VAD can suppress packets containing silence. Instead of sending VoIP packets of silence, VoIP gateways interleave data traffic with VoIP conversations to more effectively use network bandwidth. Table 2-7 illustrates the type of bandwidth savings VAD offers.

Session Initiation Protocol

No discussion about IP telephony is complete without covering the Session Initiation Protocol (SIP). SIP is a signaling protocol used to establish sessions in an IP network. A session can be something as complex as a collaborative voice video data conversation, or something as simple as a phone call. The IPT market has adopted SIP as the protocol of choice for long-term strategic direction. As of this writing, SIP remains a draft from the Internet Engineering Task Force, which is the organization responsible for developing and administering the methodology driving Internet communications. To this extent, SIP is still evolving as the IPT market matures.

Compressed Real Time Transport Protocol

Recall that voice conversation packets are sent using RTP, which runs on top of UDP. Thus, VoIP packets consist of two parts the voice and the headers (IP UDP RTP). The voice samples can be compressed, and therefore their size can vary depending on the codec usedfor example, a typical G.729 call has 20 bytes of voice samples. The headers, though, are always 40 bytes, which can be a significant amount of overhead.

Controlling Route and Cache Invalidation

The basis of any BGP conversation is the transport protocol connection that takes place between two neighbors. The neighbor connection itself is based on the OPEN message, which contains parameters such as the BGP version number. In addition, exchanged routing updates carry different attributes such as the metric, communities, and AS_PATH. Whenever an administrator changes attributes or policies, traditional BGP implementations require that a BGP TCP session with its neighbor be reset (broken and restarted) before the modified routing behavior will take effect.

Configuring Weighted Fair Queuing

Configurable elements are congestive discard threshold, the number of conversation or flow queues available, and the number of queues which may be reserved by RSVP. The number of conversation queues available controls how many different conversations the router will monitor. The default is 256. Again, this default is usually sufficient. You can monitor the maximum number of queues utilized using the command Input queue 0 75 0 (size max drops) Total output drops 426 Queueing strategy weighted fair Output queue 0 64 414 (size threshold drops) Conversations 0 11 (active max active) Reserved Conversations 0 0 (allocated max allocated)

Weighted Fair Queuing

The IP Precedence field has values between 0 (the default) and 7. As the precedence value increases, the algorithm allocates more bandwidth to that conversation or flow. This enables the flow to transmit more frequently. See the Packet Classification section later in this chapter for more information on weighting WFQ. In a Frame Relay network, FECN and BECN bits usually flag the presence of congestion. When congestion is flagged, the weights the algorithm uses change such that the conversation encountering the congestion transmits less frequently. Number of messages allowed in each queue. The default is 64 messages, and a new threshold must be a power of 2 in the range 16 to 4096. When a conversation reaches this threshold, new message packets are discarded. dynamic-queues (Optional) Number of dynamic queues used for best-effort conversations (that is, a normal conversation not requiring special network services). Values are 16, 32, 64, 128, 256, 512, 1024, 2048, and 4096. The default...

VBR Voice Solutions Using AAL2 Adaption

CES treats voice as being a constant stream of information encoded as a CBR stream. In reality, voice is not like this at all there are silences in conversation where one party speaks and the other listens. To address these limitations, the ATM forum defined a more advanced mechanism for the transport of voice as a variable bit rate compressed stream. This mechanism is described in the specification af-vtoa-0113.000, completed in February 1999, entitled ATM Trunking Using AAL2 for Narrowband Services. This approach combines the suppression of silence in the conversation with compression and the ability to multiplex multiple voice channels into a single VCC. This multiplexing of multiple voice channels overcomes the packetization delay issues resulting from the use of low-bit-rate voice encoding.

VoATM Buffering Build Out Delay

The reassembly mechanism of any ATM constant bit rate (CBR) data stream requires the reconstruction of the carried traffic at the destination at the correct moment in time. If an ATM cell is delayed during transmission, the segment assembly reassembly (SAR) function might under-run (having no data to process), resulting in gaps in conversation.

Lowerpriority classes use CBWFQ

For CBWFQ, the weight for a packet belonging to a specific class is derived from the bandwidth that you assigned to the class when you configured it. Therefore, the bandwidth assigned to the packets of a class determines the order in which packets are sent. All packets are serviced fairly based on this internal weight no class of packets may be granted strict priority. This scheme poses problems for voice traffic, which is largely intolerant of delay, especially variation in delay. For voice traffic, variations in delay introduce irregularities of transmission heard as jitter in the conversation. LLQ reduces the jitter in voice conversations. To enqueue real-time traffic to a strict-priority queue, you configure the priority command for the class after you specify the named class within a policy map. (Classes to which the priority command is applied are considered priority classes.) Within a policy map, you can give one or more classes priority status. When multiple classes within a...

Classification and Marking

Another way that could be used to reduce the bandwidth required by voice calls is voice activity detection (VAD). Recall that, on average, about 35 percent of a call is in fact silence when VoIP is used, this silence is packetized along with the conversation. VAD suppresses the silence, so instead of sending IP packets of silence, only IP packets of conversation are sent. The network bandwidth is therefore being used more efficiently and effectively.

Encapsulation Techniques

The second benefit of the encapsulated Voice over ATM approach is that it is a statistical VBR mechanism. When the conversation stops, most Voice FRADS or VoIP devices stop generating frames or packets. This directly translates into savings at the ATM layer because no cells are generated for transport during periods of silence.

Voice Activity Detection

In normal voice conversations, someone speaks and someone else listens. Today's toll networks contain a bidirectional, 64,000 bit per second (bps) channel, regardless of whether anyone is speaking. This means that in a normal conversation, at least 50 percent of the total bandwidth is wasted. The amount of wasted bandwidth can actually be much higher if you take a statistical sampling of the breaks and pauses in a person's normal speech patterns.

Network Address Translator

If Hosts A and B want to reach a destination outside the company, the NAT device will convert the source IP addresses of the packets according to a predefined (or dynamic) mapping in the NAT table on the device. Packets from Host A will reach an outside destination as coming from source IP address 128.213.x.y. Hosts in the global domain will be unaware that the address translation is taking place, and they will reply to the global address. Packets returning from an outside host will have the destination address of the packets mapped back to the private IP address of the internal source host of the conversation.

Step 1 Configure Call Transfer and Call Forward

From the analog phone, dial one of the IP phones. When the IP phone starts ringing, press the Answer softkey button followed by the Trnsfer softkey button. A dial tone emits. Enter the extension of the other IP phone. The other IP phone should ring. On the other IP phone (the target IP phone) press the Answer softkey. Notice that the call is not automatically transferred and the only connection where a conversation can occur is between the two IP phones. In fact, the caller on the analog phone is on hold, and even if the analog phone is hung up, the call transfer between the two IP phones continues.

Displays interface delays including the activated queuing mechanism with the summary information

Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 4 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) 5 minute input rate 18000 bits sec, 8 packets sec 5 minute output rate 11000 bits sec, 9 packets sec rest deleted The sample output in this figure shows that there are currently no packets in the WFQ system. The system allows up to 1000 packets (hold-queue limit) with a CDT of 64. WFQ is using 256 queues. The maximum number of concurrent flows (conversations, or active queues) is four.

Performing Call Signaling over Digital Voice Ports

Enterprise networks often use digital circuits, in contrast to analog circuits, when interconnecting their Voice over IP (VoIP) network to traditional telephony environments, such as the public switched telephone network (PSTN) or a private branch exchange (PBX). One major advantage of using digital circuits is the economies of scale made possible by transporting multiple conversations over a single circuit. For example, a digital T1 circuit using Channel Associated Signaling (CAS) (which is described in this chapter) can carry 24 simultaneous voice conversations on a single circuit.

Distribut ed Weighted Fair Queui ng

Distributed Weighted Fair Queuing (dWFQ) is a congestion management mechanism that operates FlexWAN's VIPs and provides fair treatment for outbound flows on congested interfaces. WFQ is en default on all interfaces operating at 2.048 Mbps and less. WFQ protects low-bandwidth traffic flow ensuring high volume conversations do not monopolize the available bandwidth. The mechanism w assigning each traffic flow to its own queue. A flow is defined as packets possessing the same souri address, destination IP address, source TCP or UDP port, destination TCP or UDP port, protocol ID, of service value. Once a packet is assigned to a queue, the WFQ mechanism services each queue e each queue receives a fair share of the available bandwidth, based on the assigned weight for that

Class Based Weighted Fair Queuing

CBWFQ adds a level of administrator control to the WFQ process. The same WFQ dispatch process is followed. The difference now is that the administrator can control how packets are divided into the conversations or flows. CBWFQ allows the administrator to define the various flows needed to classify traffic into unique conversations. In addition to this separation of traffic, CBWFQ can be used to guarantee that flows receive adequate bandwidth (defined by the administrator).

It Seemed Like a Good Idea at the Time

This example is important because most companies spend a great deal of time trying to control expenses in their call center environments that deal with external callers, such as customers, partners, or suppliers. However, most of the conversations I've had with companies regarding their internal support desks have a different tone. Often, they are smaller, of course. However, with large, multilocation retail firms, they can be quite large, and their approach often becomes a double-edged sword.

Link Fragmentation and Interleaving LFI

Before discussing the specifics of link-efficiency mechanisms, it seems wise to discuss the need for efficiency in the first place. Primarily, when conversations take place about using VoIP, everyone ju up and screams about the need for QoS or, more specifically, prioritization of traffic. Although prioritization is certainly necessary in a VoIP environment, lower-speed links (those under 768 kbp create problems for VoIP that simple prioritization cannot solve. In these cases, link-efficiency mechanisms are necessary to ensure proper QoS for the VoIP traffic.

Security Associations

SAs are only good for one direction of data across an IPSec connection. Because SAs are simplex, establishing conversations between peers requires two IPSec SAs, one going and one coming, for each peer and two underlying IKE SAs. IPSec SAs are also protocol specific. If you are going to be using both AH and ESP between security pairs, you need separate SAs for each.

Configuration Example Globally Modifying OSPF Interface Costs

Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 1 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 1 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated)

Scenario 123 Part C Verification and Questions

Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 0 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) Available Bandwidth 42 kilobits sec 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec 232 packets input, 17750 bytes, 0 no buffer Received 1 broadcasts, 0 runts, 0 giants, 0 throttles 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort 225 packets output, 12563 bytes, 0 underruns 0 output errors, 0 collisions, 4 interface resets 0 output buffer failures, 0 output buffers swapped out Queueing strategy weighted fair Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 1 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) Available Bandwidth 1158 kilobits sec 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec 0 packets input, 0 bytes, 0 no buffer Received 0 broadcasts, 0 runts, 0...

Scenario 122 Part C Verification and Questions

Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 1 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) Available Bandwidth 1158 kilobits sec 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec 0 packets input, 0 bytes, 0 no buffer Received 0 broadcasts, 0 runts, 0 giants, 0 throttles 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort 0 packets output, 0 bytes, 0 underruns 0 output errors, 0 collisions, 5 interface resets 0 output buffer failures, 0 output buffers swapped out 0 carrier transitions

Effects of VAD on Bandwidth

On average, an aggregate of 24 calls or more may contain 35 percent silence. With traditional telephony voice networks, all voice calls use 64-kbps fixed-bandwidth links regardless of how much of the conversation is speech and how much is silence. In Cisco VoIP networks, all conversations and silences are packetized. VAD suppresses packets of silence. Instead of sending VoIP packets of silence, VoIP gateways interleave data traffic with VoIP conversations to more effectively use network bandwidth. VAD is enabled by default for all VoIP calls. VAD reduces the silence in VoIP conversations but it also provides comfort noise generation (CNG). Because you can mistake silence for a disconnected call, CNG provides locally generated white noise to make the call appear normally connected to both parties.

Choosing a Traffic Prioritization Method

WFQ needs some method to distinguish SNA interactive traffic (encapsulated in TCP IP) from other IP traffic during periods of extreme congestion. One method that can be used is to modify the weight of the DLSw+ traffic by using the precedence bits in the IP header. When WFQ determines the packet scheduling order, the lower the weight, the higher the packet priority. The computed weight is a function of the frame length (or transmission time of completion) and its position in its conversation queue. The length of the packet is reduced using the precedence bits in the IP packet ToS field. Thus, the only advantage SNA can hope to have over other traffic lies in gaining priority (lower weight) from the precedence bits. There are several mechanisms used to set precedence on SNA traffic.

Bandwidth Considerations

On average, about 35 percent of calls are silence. In traditional voice networks, all voice calls use a fixed bandwidth of 64 kbps regardless of how much of the conversation is speech and how much is silence. When VoIP is used, this silence is packetized along with the conversation. VAD suppresses packets of silence, so instead of sending IP packets of silence, only IP packets of conversation are sent. Therefore, gateways can interleave data traffic with actual voice conversation traffic, resulting in more effective use of the network bandwidth.

Answers to the Chapter 3 QA Section

Layer 5 (session layer) controls the conversation between two endpoints. Although the term used is session, the term conversation more accurately describes what is accomplished. The session layer ensures that not only communication but also useful sets of communication between endpoints is accomplished.

What Is Network Analysis Module

Step 2 In the NAM, you can configure monitoring parameters, such as statistics, host conversation, and application response time. Figure 21-3 shows a screen capture using NAM to monitor a conversation on a particular Figure 21-3 NAM Screen Capture for Conversation on a VLAN Figure 21-3 NAM Screen Capture for Conversation on a VLAN Figure 21-4 NAM Screen Capture for Conversation and Server Response Figure 21-4 NAM Screen Capture for Conversation and Server Response Figure 21-5 NAM Screen Capture for Packet Decoder for a Particular Conversation

WAN Transport Technologies

Dwdm Concentrator

The base channel bandwidth is 64 kilobits per second (kbps), also known as digital signal level 0 (DS0). 64 kbps is the bandwidth required for an uncompressed digitized phone conversation. example, Packet A might be from an FTP session, whereas Packet B is from a voice conversation. Without MPLS, these packets would take the same route through the network. Code Division Multiple Access (CDMA) CDMA is a spread-spectrum technology that assigns a code to each conversation individual conversations are encoded in a pseudo-random digital sequence.

Classification for Voice Packets into LLQ

As you learned in previous chapters, LLQ is one of the most important Cisco QoS mechanisms to ensure quality for voice conversations, because it prioritizes voice packets over data packets at the router egress interface. For this to work, voice packets must be classified such that they are placed in the priority queue (PQ) portion of LLQ.

Call Control and Transport Protocols

Voice communication over IP is a mix of call control signals and voice conversations coded and possibly compressed into IP packets. Both reliable (connection-oriented) and so-called unreliable (connectionless) transmissions are required for voice communication. The User Datagram Protocol (UDP), which provides best-effort delivery, supplies connectionless transmission in the IP stack and is used for voice conversation transport between two endpoints. However, voice conversation loses its value with time. If a voice packet is delayed, it loses its relevance to the end user, and retransmitting it is not useful. Therefore, voice conversation uses the more efficient connectionless transport (using UDP). The following sections detail the protocols used for voice conversation traffic and call control functions. Voice Conversation Protocols Because of the time-sensitive nature of voice transport, UDP is the logical choice for carrying voice. However, voice conversation needs more information...

Example 1012 Backup Interface Is Put Back in Standby Mode

Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 1 16 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec 83 packets input, 391 bytes, 0 no buffer Received 1 broadcasts, 0 runts, 0 giants, 0 throttles 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort 90 packets output, 531 bytes, 0 underruns 0 output errors, 0 collisions, 5 interface resets 0 output buffer failures, 0 output buffers swapped out 2 carrier transitions

Introducing the macof Tool

If your LAN switch learns those frames, consider having a conversation with the switch's vendor. That being said, macof is essentially a brute-force tool and, as such, it does not embarrass itself by abiding official IEEE standards. It generates both valid and illegitimate

Protecting Against Eavesdropping Attacks

Eavesdropping attacks are also known as phone tapping attacks. The main goal is for an attacker to listen, copy, or record a conversation. An example of an eavesdropping attack is an incident reported back in 2006. The phones of about 100 Greek politicians and offices (including the U.S. embassy in Athens and the Greek prime minister) were compromised by a malicious code embedded in Vodafone mobile phone software. The attackers tapped into their conference call system. Basically, by using several prepaid mobile phones, the attackers joined the conference call and recorded their conversations. Media encryption using Secure Real-Time Transport Protocol (SRTP) delivers protection by encrypting the voice conversation, rendering it unintelligible to internal or external eavesdroppers who have gained access to the voice domain. Designed for voice packets, SRTP supports the AES encryption algorithm and is an Internet Engineering Task Force (IETF) RFC 3711 standard. Media encryption on Cisco...

Configuring Ether Channel Load Balancing

Server's address (the destination IP address) always will remain constant in the many conversations. This can cause one link to be overused if the destination IP address is used as a component of a load-balancing method. In the case of a four-link EtherChannel, perhaps two of the four links are overused. Configuring the use of MAC addresses, or only the source IP addresses, might cause the distribution to be more balanced across all the bundled links.

Forcing an Excessive Flooding Condition

Most Cisco switches do not overwrite an existing entry in favor of a more recent one however, after an existing entry ages out, a new one replaces it. Other switches function in a circular-buffer fashion when nearing full bridging-table capacity. This means that a new entry (MAC address Z, for example) simply overwrites an existing older entry (MAC address B, for example). Traffic destined to MAC address B now gets flooded out by all the ports that are members of the sender's VLAN. If a hacker constantly maintains a full bridging table, he can effectively transform the switch into a hub, which makes it easy for anyone off any port to collect all traffic exchanged in the port's VLAN, including one-to-one unicast conversations, as Figures 2-4 and 2-5 show.

Forward Error Correction

A relatively high error rate can usually be tolerated during voice conversations because the human ear might not even detect the errors. Additional consideration of minimized latency versus error correction might be required when analog data or fax is also running on the voice channel. Conversely, greater latency (delay) is not particularly detrimental to data transmission. Increasing latency does not usually reduce the transmission speed (throughput), but it can incur some additional processor overhead. Again, effective FEC (partially resulting from increased interleaving) can contribute significantly to achieving maximum throughput in a noisy environment, which produces higher throughput than when no interleaving is used.

Alarm and Event Filtering

Let us first turn our attention to filtering, not just of alarms, but of events in general. To focus an operator's or a management application's attention on those events that really matter, it is important to block out as many irrelevant or less important events as possible. This is analogous to the way in which the human brain is able to deal with the massive flow of data that it is constantly exposed to, such as sounds, visual images, and sensory data. To focus, it filters out massive amounts of data that would otherwise be distracting, for example, background noise when following a conversation.

Distributing Traffic in Ether Channel

A conversation between two devices always is sent through the same EtherChannel link because the two endpoint addresses stay the same. However, when a device talks to several other devices, chances are that the destination addresses are distributed equally with 0s and 1s in the last bit (even and odd address values). This causes the frames to be distributed across the EtherChannel links. However, if one pair of hosts has a much greater volume of traffic than the other pair, one link in the channel will be used much more than the other. This still can create a load imbalance. To remedy this condition, you should consider other methods of hashing algorithms for the channel. For example, a method that uses the source and destination addresses along with UDP or TCP port numbers can distribute traffic much differently. Then, packets are placed on links within the bundle based on the applications used within conversations between two hosts.

Debug serial interface Command

MTU 1500 bytes, BW 115 Kbit, DLY 20000 usee, rely 255 255, load 1 255 Encapsulation HDLC, loopback not set, keepalive set (10 sec) Last input 00 00 02, output 00 00 00, output hang never Last clearing of show interface counters never Input queue 0 75 0 (size max drops) Total output drops 0 Queuing strategy weighted fair Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 1 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec 3870 packets input, 206261 bytes, 0 no buffer Received 1524 broadcasts, 0 runts, 0 giants, 0 throttles 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort 3907 packets output, 228500 bytes, 0 underruns 0 output errors, 0 collisions, 174 interface resets 0 output buffer failures, 0 output buffers swapped out 24 carrier transitions

Physical DS0 Limitation

Physical DS0 limitation is a design methodology that limits the number of physical DS0 connections into the gateway. This limitation, in conjunction with other queuing methods, ensures that the gateway can successfully provide IP bandwidth across the WAN for each voice conversation originating from the individual DS0 trunks. Protects the quality of voice conversations on the WAN link by limiting the number of voice conversations that are allowed

Advanced Settings Tool

Administration - Full synchronization of Bridge subscribers with Octel node direclc Administration - Make creation of NT accounts optional Administration - Set Prefix lor TRAP dials Administration - Set the minimum extension length for locations Administration - Set the number of AA sessions allowed Administration Show hidden subscribers Audio - AGC Adjust volume target level foi new recordings Audio - AGC Gain Threshold Audio - AGC Minimum dB threshold Audio - AGC sample size Audio - Enable G72S codec support Audio Set WoveGoin dB adjustment value for playback Audio - Set WaveGain dB adjustment value foi recordings Conversation - Add dual name confirmation Conversation - Confirm Delete When Deleting a New or Saved Message Conversation - Directory List Selection Style Conversation - Disable Identified Subscriber Messaging system wide Conversation - Do not replay message summary Conversation - Exclude all receipts from voice mail stack (Exchange only) Conversation - Exclude return...

Example 22 show controllers and show interfaces with serial 0 Unplugged

Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 2 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec 390 packets input, 22659 bytes, 0 no buffer Received 367 broadcasts, 0 runts, 0 giants, 0 throttles 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort 389 packets output, 23296 bytes, 0 underruns

Voice over Frame Relay Voice Bandwidth

In a Voice over Frame Relay (VoFR) network, the frame-relay voice-bandwidth command is used in a Frame Relay map class to set aside the bandwidth required to successfully transport the desired number of calls. This method of bandwidth provisioning operates in much the same way as IP RTP Priority and Low Latency Queuing features that reserve bandwidth for traffic flows. Unlike LLQ or RTP Priority, the frame-relay voice-bandwidth command has the capability to provide CAC. Because VoFR operates at Layer 2, Frame Relay headers can be examined to determine whether a frame is carrying voice payload or data payload. The channel identification (CID) in the voice frames is used to identify which individual frames belong with the current voice conversations in progress. Because the frame-relay voice-bandwidth command sets aside a maximum bandwidth for voice conversation, and tracks the number of conversations in progress, the frame-relay voice-bandwidth command has the capability to deny the...

Securing a VoIP Network

To understand this concept, consider Figure 9-5. In the first step, the Cisco IP Phone uses SCCP to initiate a call to the PSTN. SCCP, which uses TCP port 2000, is used to communicate between the Cisco IP Phone and the UCM server. UCM determines, based on the dialed digits, that the call needs to be sent out the H.323 gateway. In the second step, using TCP port 1720, UCM initiates a call setup with the H.323 gateway. The firewall between these devices is configured to permit the H.323 protocol. The firewall is also instructed to inspect H.323 traffic, to dynamically determine which UDP ports are selected for the voice path. In the third step, UDP ports 20,548 and 28,642 were randomly selected. Because an RTP flow is unidirectional, two UDP ports are selected to support bidirectional communication. Because the firewall inspected H.323 and dynamically learned the UDP ports to be used, the firewall permits the bidirectional RTP flow for the duration of the conversation.

Bandwidth Engineering

The G.711 codec specification carries an uncompressed 64-kbps payload stream, known in the traditional telephony world as pulse code modulation (PCM). G.711 offers toll-quality voice conversations at the cost of bandwidth consumption. The G.711 codec is ideally suited for the situation in which bandwidth is abundant and call quality is the primary driver, such as in LAN environments. As you learned in previous chapters, several other features play a role in determining the bandwidth requirement of a voice call, including header compression, Layer 2 headers, and voice samples per packet. Voice Activation Detection (VAD) can also play a role in the bandwidth required by each call. VAD can be used to reduce the packet payload size by transmitting 2 bytes of payload during silent times rather than the full payload size. For example, the payload on a single G.711 packet using Cisco defaults is 160 bytes. VAD can reduce the size of the payload to 2 bytes during silent times in the...

RSVP and Weighted Fair Queuing

Inside a Cisco router, RSVP communicates with WFQ and informs WFQ of the reservations and the QoS promised in those reservations. Recall from Chapter 4 that in WFQ terminology, session's application flows (TCP UDP flows, for instance) are called conversations. RSVP reservations for sessions have a direct one-for-one relationship to WFQ conversations. RSVP also informs WFQ of a conversation queue number (which uniquely identifies a WFQ queue to use for RSVP) and the weight to use on that queue when it establishes a reservation, as illustrated in Figure 5-6. Conversation manages conversation queues manages conversation queues NOTE The actual mechanics of WFQ with RSVP are the same as described in Chapter 4 that is, each conversation has its own queue, and the weight determines QoS for the queue. The difference here is that RSVP, rather than IP prcccdcnce, determines the weight for a queue.

Example 1018 Output of show interface Commands on the Interfaces

Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 2 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec 5458 packets input, 365859 bytes, 0 no buffer Received 2900 broadcasts, 0 runts, 0 giants, 0 throttles 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort 5472 packets output, 364550 bytes, 0 underruns 0 output errors, 0 collisions, 2 interface resets 0 output buffer failures, 0 output buffers swapped out 5 carrier transitions Conversations 0 0 16 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) Conversations 0 1 16 (active max active max total) ides the Reserved Conversations 0 0 (allocated max allocated)

CUCM Signaling and Media Paths

After a call has been set up, media exchange occurs directly between the Cisco IP Phones across the IP network, using the Real-Time Transport Protocol (RTP) to carry the audio. CUCM is not involved in a call after the call has been set up. If the CUCM server were unplugged during the duration of the call, users would not notice unless they attempted to use a feature on the phone. CUCM is involved only in call setup, teardown, and features. If the CUCM server that set up the call were down during a conversation, end users would see a message indicating CM Down, Features Disabled on the LCD screen of the IP phone.

Intersite Message Passing and Directory Replication in Exchange

After you set up your messaging connector(s), you set up the appropriate directory replication connectors based on your organization topology. Since the directory replication connector relies on a messaging connector to send its directory replication messages, it can only be set up after a messaging connector is installed. Directory replication between sites is scheduled to occur every three hours by default, but may be accelerated. Upon initial installation of the directory replication connector, directory replication occurs within a few minutes and typically does not need to be accelerated. After the initial replication, only changes are sent on subsequent updates of the directory. Once directory replication has completed, subscribers can address messages to anyone in the organization by accessing either the Cisco Unity telephone directory conversation or the Exchange-based global address list (GAL).

Scenario 121 Part C Verification and Questions

Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 0 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) Available Bandwidth 1158 kilobits sec 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec 1242 packets input, 98477 bytes, 0 no buffer Received 898 broadcasts, 0 runts, 0 giants, 0 throttles 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort 1249 packets output, 91395 bytes, 0 underruns 0 output errors, 0 collisions, 2 interface resets 0 output buffer failures, 0 output buffers swapped out Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 1 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) Available Bandwidth 1158 kilobits sec 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec 1654 packets input, 90385 bytes, 0 no buffer Received 1644 broadcasts, 0 runts, 0 giants, 0...

Mpls Wan Connectivity

With the history lesson done, the conversation now moves to MPLS. Simply put, MPLS extends Layer 3 natively across the distance between central, branch, and SOHO sites. The MPLS network, though owned by the service provider, is an extension of the enterprise network. Picture the entire WAN, which was previously a Layer 2 obstacle, as a single router with multiple interfaces. It contains a routing table with all of the route entries of the enterprise network. The WAN provides any-to-any connectivity between sites without the hassle of administering a large number of circuits. Like any routed network with diverse paths, the MPLS network converges dynamically, supports multiple routing protocols, and honors QoS traffic tags and policies.

Gatekeeper Zone Bandwidth Summary

Gatekeepers do not share database information. If the primary gatekeeper fails, a secondary gatekeeper can begin to perform CAC for the network however, the secondary gatekeeper has no knowledge of the amount of bandwidth currently used in the zone or the true number of active voice conversations. Until the secondary gatekeeper has an accurate count of current calls and consumed bandwidth, the network may become oversubscribed with voice conversations. If alternate gatekeepers are used as the redundancy method, this problem is circumvented.

Analyzing Network Efficiency

Chapter 2 talked about the importance of using maximum frame sizes to increase network efficiency. Bandwidth utilization is optimized for efficiency when applications and protocols are configured to send large amounts of data per frame, thus minimizing the number of frames and round-trip delays required for a transaction. The number of frames per transaction can also be minimized if the receiver is configured with a large receive window allowing it to accept multiple frames before it must send an acknowledgment. The goal is to maximize the number of data bytes compared to the number of bytes in headers and in acknowledgment packets sent by the other end of a conversation.

Cowperation Between WFQ and QoS Signaling Technologies

As mentioned previously, WFQ is IP precedence-aware that is, it is capable of detecting higher-priority packets marked with precedence by the IP forwarder and can schedule them faster, providing superior response time for this traffic. This is the weighted portion of WFQ. The IP Precedence field has values between 0 (the default) and 7 (6 and 7 are reserved and normally are not set by network administrators). As the precedence value increases, the algorithm allocates more bandwidth to that conversation to make sure that it is served more quickly when congestion occurs. WFQ assigns a weight to each flow, which determines the transmit order for queued packets. In this scheme, lower weights are provided more service. IP precedence serves as a divisor to this weighting factor. For instance, traffic with an IP Precedence field value of 7 gets a lower weight than traffic with an IP Precedence field value of 3, and thus, has priority in the transmit order.

Logging Access List Activity

Sometimes it's useful to log the activity of traffic that is denied by an extended access list. This is especially true when you are using an access list for security and intentionally blocking conversations that shouldn't be happening. To log when packets match an access list rule, simply append the log keyword to the rule (logging is not available on standard access lists). For example, to log every time a packet is denied from sources matching 172.16.1.0 0.0.0.255, the access list might look like this

Example 44 show interface Output

Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 2 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec 179 packets input, 12647 bytes, 0 no buffer Received 70 broadcasts, 0 runts, 0 giants, 0 throttles 1 input errors, 0 CRC, 1 frame, 0 overrun, 0 ignored, 0 abort 173 packets output, 17321 bytes, 0 underruns 0 output errors, 0 collisions, 78 interface resets 0 output buffer failures, 0 output buffers swapped out 106 carrier transitions

Monitoring and Troubleshooting RSVP

BPS To From Protoc DPort Sport Weight Conversation 44K 145.20.0.202 145.10.0.201 UDP 1000 1000 0 264 44K 145.20.0.202 145.10.0.201 UDP 1001 1001 13 266 98K 145.20.0.202 145.10.0.201 UDP 1002 1002 6 265 1K 145.20.0.202 145.10.0.201 UDP 10 10 0 264 RSVP Serial3 0 has no installed reservations Router Conversation supports 1 reservations Data given reserved service 316 packets (15800 bytes) Data given best-effort service 0 packets (0 bytes) Reserved traffic classified for 104 seconds Conversation supports 1 reservations

The Neighbor State Machine

The initial state of a neighbor conversation indicates that no Hellos have been heard from the neighbor in the last RouterDeadInterval. Hellos are not sent to down neighbors unless those neighbors are on NBMA networks in this case, Hellos are sent every PollInterval. If a neighbor transitions to the Down state from some higher state, the link state Retransmission List, Database Summary List, and link state request list are cleared.

The Neighbor Data Structure

An OSPF router builds the Hello packets for each network using the information stored in the interface data structure of the attached interface. By sending the Hello packets containing this information, the router informs neighbors about itself. Likewise, for each neighbor the router will maintain a neighbor data structure consisting of the information learned from other routers' Hello packets. This two-way exchange of information with a neighbor can be thought of as a conversation.

Step 3 Specifying Static Routes

MTU 1500 bytes, BW 64 Kbit, DLY 20000 usec, rely 255 255, load 1 255 Encapsulation PPP, loopback not set Last input 0 00 06, output 0 00 06, output hang never Last clearing of show interface counters never Input queue 0 75 0 (size max drops) Total output drops 0 Output queue 0 64 0 (size threshold drops) Conversations 0 1 (active max active) Reserved Conversations 0 0 (allocated max allocated) 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec 359 packets input, 5814 bytes, 1 no buffer Received 0 broadcasts, 0 runts, 0 giants

Scenario 61 IP Addressing and Subnet Calculation

Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 0 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) Available Bandwidth 1158 kilobits sec Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 0 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) Available Bandwidth 1158 kilobits sec 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec 125 packets input, 7634 bytes, 0 no buffer Received 124 broadcasts, 0 runts, 0 giants, 0 throttles 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort 161 packets output, 9575 bytes, 0 underruns 0 output errors, 0 collisions, 1 interface resets 0 output buffer failures, 0 output buffers swapped out 4 carrier transitions DCD up DSR up DTR up RTS up CTS up Ethernet0 is up, line protocol is up

Understanding Voice VLANs

It is a common and recommended practice to separate voice and data traffic by using VLANs. There are already easy-to-use applications available, such as Wireshark and Voice Over Misconfigured Internet Telephones (VOMIT), that allow intruders to capture voice conversations on the network and convert them into WAV data files. Separating voice and data traffic using VLANs provides a solid security boundary, keeping data applications from reaching the voice traffic. It also gives you a simpler method to deploy QoS, prioritizing the voice traffic over the data.

Understanding RTP and RTCP

Once two devices attempt to establish an audio session, RTP engages and chooses a random, even UDP port number from 16,384 to 32,767 for each RTP stream. Keep in mind that RTP streams are one-way. If you are having a two-way conversation, the devices will establish dual RTP streams, one in each direction. The audio stream stays on the initially chosen port for the duration of the audio session (the devices will not dynamically change ports during a phone call).

Understanding Digital Connections

As an office grows, analog connectivity can become cumbersome and expensive. Digital connections allow multiple conversations to share a single circuit. The digital channels are built using DS0 channels as building blocks. Depending on the interface you are using to connect to the PSTN service provider, you can support a different number of DS0 channels. Each DS0 provides a 64-kbps channel, which supports a single audio call. The following are the available digital ports you can use in your router

Step 5 Subtract Bandwidth Savings Measures

VAD allows the router to detect the sound of silence in a VoIP conversation. By default, the routers will send RTP data, even if no one is talking. Through many studies, findings show that on average, 35-40 percent of a phone call is silence. By enabling VAD, you are able to recoup this bandwidth back into your budget. RTP header compression (often called compressed RTP) allows routers to cache network and transport layer header information in voice packets, allowing routers to strip this redundant header information from future packets. Think about it this way once two devices establish a VoIP call between each other, what type of information appears in the network and transport layer headers IP addresses, port numbers, QoS markings, and so on. Now, how much of that information changes during the course of a voice conversation None of it. The IP addresses stay the same, the port numbers don't change, and the QoS markings are consistent. So, rather than send this redundant information...

Live Reply and Live Record

The live record feature enables a subscriber to record a live call and have that call delivered into the subscriber's mailbox. For example, John calls Bill and decides that he needs to record this conversation. John can invoke the live record feature and his conversation with Bill will be stored in John's mailbox. Once the message has been delivered to the mailbox, it can be played, replayed, saved, or forwarded to any subscriber or group of subscribers.

Scenario 51 IP Addressing and Subnet Calculation

Conversations 0 1 64 (active max active threshold) Reserved Conversations 0 0 (allocated max allocated) 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec 27 packets input, 2452 bytes, 0 no buffer Received 27 broadcasts, 0 runts, 0 giants, 0 throttles 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort 29 packets output, 2044 bytes, 0 underruns 0 output errors, 0 collisions, 28 interface resets 0 output buffer failures, 0 output buffers swapped out 7 carrier transitions DCD up DSR up DTR up RTS up CTS up Serial1 is up, line protocol is up Hardware is HD64570 Internet address is 199.1.1.97 27 Conversations 0 1 64 (active max active threshold) Reserved Conversations 0 0 (allocated max allocated) 5 minute input rate 0 bits sec, 0 packets sec

Overview of Multilayer Switching

Strictly defined, a flow is a specific conversation, consisting of multiple packets, between a network source and destination within a specific time sequence. Let's take a user that is pulling down a web page from a specific web server. This example would be one flow. The same user could be performing a File Transfer Protocol (FTP) file transfer at the same time from an FTP server. This example would be a completely different flow. Two different applications two different protocols two different flows however, only one host is performing two flows. In terms of flows, there is no distinction between unicasts or multicasts.

Foundation Topics InterVLAN Routing Background

Routing in a switch network can take on three forms interVLAN routing, multilayer switching (MLS), and Cisco Express Forwarding (CEF). InterVLAN routing is based on adding a route processor somewhere in the switched network to provide Layer 3 routing. Every packet destined from one VLAN to another must pass through the router. MLS, on the other hand, is based on the principle that the router only sees the first packet of a conversation. Switching paths are then set up so that subsequent packets bypass the router and are switched by a more efficient shortcut path. CEF is a distributed switching mechanism keeping copies of route cache information in several different forms to be used for efficient switching. Catalyst switches can hand off packets to a CEF-capable router for processing, as in interVLAN routing. Some Catalyst platforms implement CEF directly in hardware.

Show interface serialinterface Command

MTU 1500 bytes, BW 115 Kbit, DLY 20000 usec, rely 255 255, load 1 255 Encapsulation HDLC, loopback not set, keepalive set (10 sec) Last input 00 00 02, output 00 00 00, output hang never Last clearing of show interface counters never Input queue 0 75 0 (size max drops) Total output drops 0 Queuing strategy weighted fair Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 1 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec 3870 packets input, 206261 bytes, 0 no buffer Received 1524 broadcasts, 0 runts, 0 giants, 0 throttles 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort 3907 packets output, 228500 bytes, 0 underruns 0 output errors, 0 collisions, 44 interface resets 0 output buffer failures, 0 output buffers swapped out 24 carrier transitions

Scenario Part C Verification and Questions

Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 0 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) Available Bandwidth 1158 kilobits sec 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec 1242 packets input, 98477 bytes, 0 no buffer Received 898 broadcasts, 0 runts, 0 giants, 0 throttles 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort 1249 packets output, 91395 bytes, 0 underruns 0 output errors, 0 collisions, 2 interface resets 0 output buffer failures, 0 output buffers swapped out 12 carrier transitions DCD up DSR up DTR up RTS up CTS up Serial1 is up, line protocol is up Hardware is HD64570 Internet address is 163.1.23.202 25 Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 1 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) Available Bandwidth 1158 kilobits sec 5 minute input rate 0 bits sec, 0 packets...

Example 121 Example of show interface Command

Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 1 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec 2229 packets input, 30711 bytes, 0 no buffer Received 82 broadcasts, 0 runts, 0 giants, 0 throttles 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort 297 packets output, 4413 bytes, 0 underruns 0 output errors, 0 collisions, 645 interface resets 0 output buffer failures, 0 output buffers swapped out 1290 carrier transitions DCD up DSR up DTR up RTS up CTS up frame_switch

Connection Termination

UDP is a connectionless, best effort transport-layer protocol. UDP uses IP protocol number 17. Because UDP does not spend time setting up and handling connections, it generates less overhead than TCP. UDP's benefit is that with a small header, it can send data with less overhead to the destination. Applications that are time-sensitive, such as voice and video, tend to use UDP transport instead of TCP. There is no need to retransmit a voice conversation.

Providing TCP Load Distribution

When NET comes up in everyday conversation, you probably think of it as a translation mechanism that allows your company to access the Internet. NET has another function that is unrelated to this feature. If your company has multiple hosts that communicate with a heavily used host or server, you can use NET to establish a virtual host on the inside network that coordinates load sharing among multiple real hosts. Allocation is done on a round-robin basis from a rotary pool of real addresses when a n ew connection is opened from the outside to the inside. Eny non-TCP traffic is still passed without translation, unless other translations are in effect.

The ASA might also proxy the TCP connection TCP InterceptSYN cookies if configured

The TCP Intercept feature (PIX OS 5.2 and later) provides additional protection. Once the SYN threshold is reached, and until the SYN count falls below the minimum threshold, every incoming SYN is intercepted. For every incoming SYN at that point, the PIX will respond (on behalf of the destination station) with an empty SYN ACK. If the final ACK is received (valid three-way handshake) then the PIX will initiate with the destination station on behalf of the original sender. The TCP conversation continues as normal.

Example 67 Verifying the D Channel Subinterface

Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 1 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec 102 packets input, 618 bytes, 0 no buffer Received 0 broadcasts, 0 runts, 0 giants, 0 throttles 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort 102 packets output, 571 bytes, 0 underruns

Port Number Assignment Strategy

PAT divides the available ports per global IP Address into 3 ranges 0 - 511, 512 - 1023, and 1024 - 65535. Cisco IOS and PIX OS will attempt to assign the same port value of the original request. However, if the original source port has already been used it will start scanning from the beginning of the particular port range to find the first available port and assign it to the conversation.

Example 56 show interface async 65 Command Output

Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 1 16 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) Available Bandwidth 6 kilobits sec 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec 0 packets input, 0 bytes, 0 no buffer

Pix Acl Configuration Philosophy

Since the ASA access check applies only to the initial packet of a conversation, the access-lists are only evaluated once per connection. If the application changes port information after the initial packet exchange, the ASA will pick up on that, so no additional configuration is necessary on the PIX (unlike a classic router's ACL).

Example 112 Weighted Fair Queuing on an Interface

Output queue 6 1000 64 0 (size max total threshold drops) Conversations 2 3 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) Available Bandwidth 42 kilobits sec (depth weight total drops no-buffer drops interleaves) 1 32384 0 0 0 Conversation 31, linktype ip, length 44 (depth weight total drops no-buffer drops interleaves) 5 32384 0 0 0 Conversation 147, linktype ip, length 1376

Example 109 Primary and Backup Interfaces with Low Traffic

Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 1 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec 19 packets input, 1913 bytes, 0 no buffer Received 7 broadcasts, 0 runts, 0 giants, 0 throttles 1 input errors, 0 CRC, 1 frame, 0 overrun, 0 ignored, 0 abort 21 packets output, 3143 bytes, 0 underruns 0 output errors, 0 collisions, 2 interface resets Conversations 0 0 16 (active max active max total) Reserved Conversations 0 0 (allocated max allocated)

Example 102 Backup Interface in Standby Mode

Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 1 16 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) eaders for the CC 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 0 bits sec, 0 packets sec

Example 1011 Primary and Backup Interfaces When the Load Threshold Exceeded

Conversations 1 2 256 (active max active max total) Reserved Conversations 0 0 (allocated max allocated) 5 minute input rate 0 bits sec, 0 packets sec 5 minute output rate 392000 bits sec, 426 packets sec 124 packets input, 8885 bytes, 0 no buffer Received 63 broadcasts, 0 runts, 0 giants, 0 throttles 1 input errors, 0 CRC, 1 frame, 0 overrun, 0 ignored, 0 abort 215234 packets output, 24533109 bytes, 0 underruns 0 output errors, 0 collisions, 2 interface resets 0 output buffer failures, 0 output buffers swapped out 0 carrier transitions Output queue 0 1000 64 0 (size max total threshold drops) Conversations 0 1 16 (active max active max total) Reserved Conversations 0 0 (allocated max allocated)

What Problems Need to Be Solved

User Datagram Protocol (UDP) is an alternative protocol to TCP that also operates at Layer 4. UDP is considered an unreliable, connectionless protocol. Although unreliable may have a negative connotation, in cases where real-time information is being exchanged (such as a voice conversation), taking the time to set up a connection and resend dropped packets can do more harm than good.

It makes the PIX less stealthy but is recommended

IDENT is a method of verifying a stations name or other information. Typically it is not allowed for an incoming connection, as it is not relevant to business needs. However, a site will wait for the timeout of IDENT (if configured) before proceeding with the original protocol conversation.

How Do You Catch a Criminal When Everyone Wears a Mask

Traditional firewalls can be very effective against network intrusions in which packets are sent from an untrusted network to a trusted one. The IP protocol being used can be checked for appropriateness and to ensure that it is part of a conversation that originated from the trusted side. In this way, security attacks that rely on manipulating certain IP protocols can be halted. Increasingly, the number of IP protocols used on the network and the Internet is narrowing dramatically. With the popularity of TCP IP applications, and even more so web-based applications using HTTP, the vast majority of traffic on modern networks ends up being only a couple of IP protocols. In addition, some protocols encapsulate themselves in other IP protocols to avoid problems with firewalls. For example, IPsec may be encapsulated within UDP packets. RTP video streams may be encapsulated within TCP. Finally, when nearly every corporate application becomes a web-based application, HTTP is the predominant...

Digital Subscriber Line DSL

DSL uses the existing phone wires connected to virtually every home in most countries. The twisted-pair wires that provide phone service are ideal, because the available frequency ranges on the wires far exceed those required to carry a voice conversation. Human speech occupies frequencies of roughly 4000 hertz (4 kHz) or less. The copper wires that provide phone service can carry in the range of 1 to 2 million hertz (1 to 2 MHz). DSL provides more downstream data (from the Internet to you) than upstream data (from you

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