Different applications have different requirements. The introduction of voice and streaming video onto the existing IP network presents a completely new set of requirements to the operational performance envelope of the network. This chapter will examine the importance of assessing the readiness of the network and fully evaluating design considerations related to delivering integrated service in the enterprise. To address new service integration, the chapter will explore a methodology, called the "performance envelope," for mapping the characteristics of the corporate network.
As businesses delve into VoIP and video solutions in the enterprise, their focus is on business drivers. For many companies, cost reduction is a key business driver for convergence. Cost cannot be the only driver, and for practical purposes, shouldn't be the primary driver. It's important to set reasonable expectations and to fully understand all the business drivers behind service integration. A new application service or an integrated VoIP and CRM solution will require consideration of very different factors than cost reduction alone. It's vital to fully understand the business motivation for converging video, voice, and data so that the project planning and implementation teams can address the true success factors when mapping out project milestones. Communicating the objectives clearly with everyone involved helps maintain a clear view of the expected results.
Ensuring network readiness is a significant task and not to be taken lightly. Over the past 4 or 5 years, both VoIP vendors and systems integrators have learned the hard lesson of failure because of inadequate preparation and planning. The existing data network has been optimized over time to support the existing business requirements. As new applications and services are added, network tuning takes place and a variety of parameters are often tweaked. Historically, these optimization efforts have been driven by packet data applications such as CRM, ERP, and Web services solutions—"normal" IP applications. Voice and video present new challenges as they add streaming, real-time traffic to an existing service network. VoIP call quality requires low latency in the network. Jitter needs to be low, and to avoid complicated jitter buffering requirements, it needs to be consistent. Bandwidth and packet loss need to support the new, integrated services.
Testing and documenting the parameters and characteristics of the existing networks before extensive planning will help ensure the network is capable of supporting VoIP, video, or both. It's important to understand aspects of the traffic needed to support all the different service types that will coexist on the new network. Traffic types, frame size, prioritization schemes for quality of service (QoS), jitter, latency, and packet loss are all crucial factors. Because consistent network performance is so important, it's also prudent to evaluate utilization in the network both at peak and normal times.
If the network will include PSTN gateways for global connectivity, there may be requirements specific to the gateway demarcation between networks. Both the IP network and PSTN need to be considered. The PSTN side of the gateway will have specific trunking requirements. These have traditionally been calculated using Erlang-B formulas. The IP network must support the bandwidth and other network characteristics to ensure a clean handoff of calls from one network to the other.
Telephony carrier central offices and enterprise PBX systems are engineered for trunking requirements using the Erlang-B traffic tables and hundred calling seconds per hour, or CCS. 36 CCS would represent a circuit at maximum occupancy with zero idle time. I won't explore the formulas in any depth as we investigate managing the converged services network, but it's worthwhile to understand the complexity and sophistication of traffic engineering in the traditional telephony environment. The formula for determining voice traffic load that's most widely used is:
Erlangs are really a simple concept identifying the number of calling minutes per hour a system can handle. The key differentiatior is between the traditional circuit world, where minutes equate to busy circuits, against the VoIP world, where minutes of use may not be applicable. In VoIP services, it's really the carrying capacity of the network in packets that's the key indicator.
The enterprise IP network must complement the trunking requirements when connecting to the PSTN. Overlooking details like this may result in performance bottlenecks that impact call quality.
Whenever new technologies are implemented in the corporate network, there are some basic planning steps that should not be overlooked. These basics will improve the likelihood of success in the deployment of converged voice and video services.
Although this brief section seems to be focused on the planning and implementation of integrated services, that is the crucial starting point of holistic management of the total integrated services network. When an organization decides to implement VoIP, that decision commonly leads to acceptance of a new, holistic approach to managing the entire service network.
Before undertaking a convergence project, fully identify the business drivers for change. It's important to identify the business objectives and set everyone's expectations accordingly. List the benefits you expect to derive from integrating services so that each benefit can be addressed individually. Broad, sweeping objectives such as "save money" aren't measurable. The success of VoIP implementations really depend on advance data gathering and analysis.
The more you can document prior to any implementation work, the greater your likelihood of success. It's important to balance objectives and expectations with analysis of your network's capabilities. The determination that your network can support the new, converged services, or learning that it can't and will require upgrades in routing and switching, will influence both the budget and project timelines.
There's an old adage that says "people don't plan to fail, they fail to plan." It's vital that your project team be methodical and document every facet of the entire planning process. One widespread complaint heard from both systems integrators and VoIP vendors has been that the assessment and planning has been the weak link in a high percentage of failed projects.
Resources invested in planning, whether spent on a project manager or the hours spent by a technical team evaluating requirements, will bring benefits at cutover time. There is no substitute for preparation and planning. For an enterprise mission-critical service such as voice, failure can be especially painful, both in stress and in business impact. The more meticulous the planning process, the higher the chance of success.
Once the network requirements have been documented and are fully understood, test to make sure the existing network can support your VoIP requirements. Comprehensive analysis up front helps ensure that you know what to test and what parameters or characteristics your network requires. It's important to approach testing with the specific aim of supporting the fully converged network. Testing VoIP services alone is not enough. Existing applications should be tested while simulating VoIP. There is a hidden danger of integrating VoIP services and doing all the right things to support call quality, only to starve network resources for other mission-critical applications. It is absolutely vital to take a holistic view of all network services and applications when testing.
The "right" resources will vary from organization to organization. For some, acquiring resources might mean dedicating a project manager. Most organizations will need some training to bring existing staff up to speed on VoIP resources. For many companies, this will entail bringing in a VAR partner or systems integrator to bolster the expertise available. There is no shame in admitting you need expertise from outside and working with a consulting partner. Leverage expertise everywhere you can.
To future-proof your network and ensure it can continue to grow to support the enterprise business, a holistic approach that scales both vertically and horizontally will help enable future growth at an acceptable pace. It will enhance readiness at anticipating that next new application requirement that lies just beyond the horizon now. The one you don't yet know about. The VoIP integration project presents an opportunity to review the overall future capacity of your network.
Once you've developed all your business needs and determined that VoIP service integration is the right business move, you need to gather data to fully understand voice services requirements. In telecommunications, there are patterns of calling that can be identified. Call usage reports from either the existing PBX or the incumbent telephony services provider offer a wealth of information. Some variation of these reports should be readily available from the current telephony technology, whether it's carrier-based Centrex-like service or a locally administered PBX.
There are several key elements of voice service that need to be well understood. Business telephone systems often focus on the "busy hour," or that hour of the day the most telephone traffic occurs. This is a starting point, but there are many nuances to consider. The busy hour for inbound calls may be different than the busy hour for outbound calls. The aggregated inbound/outbound busy hour may be different than either taken individually. These busy hours may vary on different days of the week.
Most companies also have an identifiable busy day in the average week. And business managers understand the cyclical nature of their respective businesses. The busy month of the year is generally well known in any business sector.
For large enterprises with multiple divisions, or business units, calling patterns may vary and some may have different requirements and calling patters than other divisions of the corporate headquarters. Flexibility is key to understanding the needs of all business units.
To identify specific business needs, you must understand the calling flows and volumes. Some enterprise calling patterns are very inbound in nature. This might be easily identified by a company using a call center, but other companies establish large outbound call centers as well. The patterns for inbound and outbound calls are often very different. It's vital to know what is required to support your particular business. It's also important to identify upper call volumes, or absolute peak traffic.
Most enterprises are somehow cyclical in nature. When analyzing calling patterns, it's worthwhile to account for several months or even for the past year. Peak periods vary widely by industry. For example, a retail services company may experience much higher call volumes during the Christmas season than any other time of year, whereas a financial services company might expect the highest call volumes at tax preparation time approaches. It's important to take these peak activity periods into account and not simply design the VoIP services to support dayto-day business needs.
As you develop a strategy for integrating VoIP into the network, it's also a good time to recognize that the PSTN has become a mission-critical component of business operations. The
PSTN network has been engineered to deliver 99.999 percent uptime. That is what the traditional carriers tout as their benchmark. This service level equates to roughly 5 minutes of downtime per year. Corporate data networks rarely provide this level of reliability. For many organizations, increasing network uptime will provide incentive for network enhancement and redesign requirements during preparation and implementation of a converged services network.
One factor to consider in VoIP deployment is the encoding method used to convert analog human speech into a digital signal for packetized transmission. There are several encoding techniques used in VoIP solutions. Each uses different compression levels. These compression levels affect the bandwidth needed to transmit the voice packets, the quality of the voice, and the processing time. Pulse Code Modulation (PCM), or G.711 encoding, is widely used. It's the standard encoding used in the PSTN, and is generally assumed to provide the best voice quality, although new wideband codecs are now starting to surpass PCM in some implementations. PCM generates a 64Kbps voice stream. For some companies, bandwidth requirements, weighed against quality considerations, might make a different encoding scheme more appropriate. Increasing the compression of the voice signal reduces the overall bandwidth requirement, but voice quality may suffer as a result.
Even though the PSTN is more than 100 years old, voice quality is still often measured by having a group of people listen to sounds in headphones and subjectively rate the quality. These evaluators rate the sound quality on a scale of 1 to 5. This rating, from 1 to 5, is referred to as the mean opinion score (MOS). A rating of 1 roughly equates to the scratchy sound quality of the intercom speaker in a fast food drive through or a warehouse environment. A 5 is the highest rating and is considered "perfect," or the theoretically highest-grade voice quality achievable. It has always been the target for conducting corporate business calls, but as you'll see in the chart, not actually achieved. A score of 4.4, garnered by the G.711 codec used in the public telephone network, was described in the past as "toll quality" voice. This has been the traditional carriers' target for business-class voice services.
Although the statistical validity of this human sampling may be questionable to some, this approach has been used for many years and is generally accepted globally. Today, MOS is often determined using ITU Recommendation P.800, which details methods for subjective determination of transmission quality. In the real world, the human ear can clearly distinguish between a 4 MOS and a 4.5 MOS.
Table 5.1 provides a comparison of several widely adopted encoding schemes. This table describes the codec types and algorithms used, the sample size, bit rate, and encoding delay for the algorithm used. The last column identifies the MOS for these codecs. When selecting a codec for VoIP implementations, it's important to remember that delay is cumulative from end to end. Encoding delays can become real call quality concerns in a large enterprise network. If the total delay exceeds 250 milliseconds (ms), call quality will suffer. The ITU-T standards (http://www.itu.int/) provide extensive details about the operation of these codecs. Vendors and consulting partners will also be a valuable resource in identifying the best approach for each specific implementation.
ITU Codec Descriptor
Adaptive Differential Pulse Code Modulation (ADPCM)
Sub-Band Adaptive Differential Pulse Mode Modulation (SB-ADPCM)
Low-Delay Code Excited Linear Predictive (LC-CELP)
Conjugated Structure Algebraic Code-Excited Linear Predictive (CSACELP)
Algebraic Code-Excited Linear Predictive (ACELP)
Table 5.1: Encoding schemes.
PCM, or G.711, is the method that has been used in the PSTN. It's perhaps the most widely used codec in VoIP systems because it provides a well-known call quality rating. Every VoIP equipment manufacturer supports G.711. G.726, known as Adaptive Differential Pulse Code Modulation (ADPCM), may reduce the bandwidth requirements by 50 percent while only posing minimal degradation (.2 of a point) on the MOS rating.
G.722 is mostly used in FM radio, but is included here because FM radio quality is a good audio fidelity comparison. It's rarely used in VoIP solutions. G.722.1 offers lower bit-rate compressions. A more recent variant, G.722.2, also known as AMR-WB (Adaptive Multirate Wideband), offers even greater compressions and the ability to adapt quickly to varying compressions if needed due to network changes. Bandwidth is conserved during periods of high network congestion, then returned to a normal level when congestion is alleviated. Because G.722 and its variants sample audio data at a rate of 16kHz, rather than the PSTN PCM method at 8KHz, these codecs result in far better sound quality, clarity, and fidelity.
G.728, or Low-Delay Code Excited Linear Predictive (LC-CELP), coding is widely used in voicemail systems for digitizing stored voice messages. G.729 can deliver an 8-kilobit sample with less than 16ms of processing time. This G.729 codec standard is often used in Voice over Frame Relay (VoFR) and is supported by many frame relay equipment vendors.
The G.729a codec is a variation on CS-ACELP that is rapidly growing in popularity. It compresses the voice stream into 10ms frames. It's less CPU intensive that many of the others, making it a good choice for bandwidth-constrained implementations. It offers a blend of bandwidth and delay performance characteristics that are making it quite popular for VoIP services. It has the added advantage of interoperating with G.729 codecs.
G.723.1, or Algebraic Code-Excited Linear Predictive (ACELP), codecs work differently. They create models of the human voice and "predict" what the next sound will be. These codecs encode the differences between the predicted sound and the actual sound. Only this difference is transmitted to the receiving end, reducing bandwidth requirements. Although bandwidth needs may be reduced, this codec has often resulted in complaints that women and children's voices were not represented accurately.
The codecs described earlier are narrowband codecs, initially developed for supporting narrow voice channels. The PSTN voice channel is a 64kbps circuit carrying audio signals in the range from 300Hz to 3000Hz. Some pure IP voice solutions are now beginning to adopt broadband codecs, mostly proprietary, for VoIP call processing. It's expected that the use of these, and other, new broadband codecs will increase as more and more voice calls are carried over IP in the future.
As mentioned earlier, the PSTN has been in operation for well over 100 years. During that lifetime, it has been tuned and optimized to deliver one type of traffic, voice, as efficiently as possible with acceptable call quality. There are quality assurances in the traditional PSTN that are sometimes inferred, based on past use, and for enterprise business, sometimes contractually guaranteed. In short, voice service on the PSTN is a known quantity. You know exactly what you expect for call quality when you pick up the telephone.
The PSTN is a "connection oriented" network. When you make a telephone call, the network establishes a voice path, or circuit, across the network that is dedicated to that specific telephone call, and meets all the basic quality requirements you've come to expect. Over the years, the telecommunications carriers have optimized the PSTN to support the delay, jitter, and loss characteristics that guarantee the expected voice call quality. In this connection-oriented, or circuit-switched, network, each circuit is dedicated to a single conversation, so security requirements are easily met. The circuit resources aren't shared as they are in an IP network. Once the call is completed, the circuit is disconnected, or "torn down."
IP networks such as the Internet share resources all the time. They were designed to carry multiple traffic types. That's the fundamental shift being embraced with convergence. In the legacy past, we designed and optimized a specific network for each type of traffic. A large enterprise might well have a voice network, a data network, and a video network, each with different performance characteristics. Convergence is about leveraging all these technologies— data, voice, and video—onto one single network infrastructure. To succeed, it's important to understand how these different networks behave.
IP is commonly referred to as a "best efforts" protocol. The protocol itself provides routing delivery, but if delivery attempts fail, for any reason, IP will discard the packets. IP was designed to use available network resources in real-time, to deliver data, but without any assurances of delivery and with no quality guarantees. IP was initially designed to support data traffic that is inconsistent and unpredictable in volume, or "bursty." IP data is described as bursty in nature because different data types have different characteristics. The data flow for email messages or Web browsing is quite different than the data flow for streaming media such as voice or video.
IP is a "connectionless," or packet-switched, network. Dedicated circuits aren't established. Messages are divided up and placed into packets. Each packet is somewhat like an envelope in the postal system. It has header fields that contain the source address and a destination address. IP uses routing protocols to identify paths through the network. These packets take the best path available at the time, based on whatever metric a particular routing protocol uses to determine the best path. Thus, as congestion and other factors impact the performance of the network, IP packets can easily take different paths across the network, arriving at different times. A large message is often made up of many IP packets, so they need to be buffered and reassembled at the receiving end.
QoS includes characteristics such as delay, jitter, and packet loss. IP doesn't provide any assurances for quality or delivery, so performance requirements can only be achieved by adding some other feature, or protocol, to the network. In practical terms, we increase bandwidth, optimize switching and routing paths, and incorporate other, higher-layer protocols to meet VoIP call quality requirements. We also add a layer of security by incorporating industry best practices in network design, incorporating firewalls and user access controls, and including monitoring and response mechanisms throughout the network. Table 5.2 highlights important differences in characteristics of the PSTN when compared with typical IP networks.
Characteristics of the PSTN
Characteristics of IP Networks
Designed and optimized to support voice as a single traffic type
Designed to carry any type of traffic
Uses dedicated circuit for guaranteed delivery
No delivery guarantees using packet-based routing
Connection-oriented—A dedicated circuit path, or connection, is created for each voice conversation
Connectionless in nature—Conversations are digitized and inserted in many packets
Phone calls are typically long duration (4 minutes on average)
IP packets are limited to 65,536 bits; packets are small and only "live" on the network for a very short time; a conversation is made up of many packets
PSTN circuit is dedicated to one 64Kbps voice channel
The IP network uses all the available resources of the network
Delay, jitter, and loss have been designed to meet specific levels for voice calls
Delay, jitter, and loss must be overcome by adding resources to the network
Dedicated circuit provides generally acceptable security for voice calls
Security requires special resources in the network
Table5.2: Performance characteristics in the PSTN vs. IP networks.
Because IP packets are small in size and data traffic is bursty in nature, IP networks use routing protocols to pick the best path for transmission. In a typical IP network, this bursty nature of data means that the traffic load on the network is constantly changing throughout the day. This constant state of flux means that the best path identified by routing protocols may change often. Routing protocols are dynamic and constantly update to identify the best path through the network based on the current traffic load. The best path one minute may not be available the next. For VoIP calls, this means that the packets comprising a conversation can potentially take many different paths through the network.
Traditional data use of IP is often not particularly sensitive to network delay. If an email message is broken into many packets that can't be delivered immediately, there is likely no problem. The message can be placed in many different packets, then routed across any number of paths available in the network. The entire message can be reassembled and delivered to the recipient, with no repercussions. Unlike email, voice calls are real-time interactions between people. Even a half-second of network delay can be so detrimental to voice quality that VoIP service is rendered unusable.
The PSTN dedicates bandwidth, a 64Kbps voice channel, to each telephone call. It dedicates this bandwidth for the duration of the call. It does this consistently and predictably for every phone call, every time. In an IP network, all the resources of the network are available to every user or node on the network. This is necessary for routing protocols to work and so that IP can make "best efforts" at delivery. Thus, every voice call can be impacted by changing traffic load and network conditions.
Over time, we keep adding new traffic types to IP networks. Today, these networks support email, file transfer, Web surfing, and data sharing. As we add new traffic types to the network, we must redesign and modify the networks to support not just the new traffic type but all traffic types in the aggregate as well. The challenge with implementing and managing a converged VoIP service network is meeting user expectations such as call clarity and fidelity without delay or jitter impairments. Voice conversations must be clear and intelligible or uses won't trust the integrity of the network for voice calls. Beyond the quality of the voice itself, the basics of delivering dial tone, successfully completing calls, and always being available are key quality concerns.
Managing call quality perceptions is a bit like walking the razor's edge. Because voice service and telephone calling are so widely used and accepted, user expectations are high, based on their own experiences. When using a new VoIP service, if users feel that call quality is unacceptable, they will simply hang up and use alternative methods. In consumer VoIP solutions, this perception problem is a constant battle. In the enterprise environment, you must take proactive steps during the implementation process to level-set user expectations with what the integrated service can functionally support. If users develop mistrust of the network, even based on misperceptions of capabilities, it is very difficult to regain their confidence. You need to be able to measure, guarantee, and deliver suitable call quality. You also need to ensure that the network can continue to support the pre-existing data traffic without service degradation to existing services.
An often-stated (by consultants, systems integrators, and equipment vendors) misperception is that many corporate networks were poorly designed. It's simply not true. It's important to recognize the truth so that we aren't doomed to repeat the mistakes of the past.
Existing networks in business were not poorly designed, but in many cases, they weren't designed at all. For many organizations, PCs were implemented way back in the past. Windows 95 brought in the concept of the workgroup, and local area networks (LANs) emerged, as small, isolated pockets of information, typically for a workgroup. With routing and internetworking, organizations began interconnecting LANs, and the explosive growth of the Internet was just one more connection.
Today, we have business partner connections, VPN links, Web services, and more. We have the Internet, the intranet, and extranets. We have Web servers, offering Web services, supporting ebusiness. And in many cases, each new feature or function was brought about by tweaking the service network with some incremental upgrade. We didn't design a large number of our networks, they just happened.
Today's networks have evolved and grown over time. Many have been carefully and methodically redesigned every few years. Like the PSTN has been tuned and optimized for voice traffic, there are IP networks that have been designed and tuned to support known and understood data applications that exist within the enterprise. The enterprise applications in use today may have very different network performance requirements than converged video and/or VoIP services.
VoIP projects begin with an implied assumption that some network upgrades may be necessary. Equipment in the network, such as routers, may already be running at high CPU utilization. In many cases, the router OSs may not be able to support VoIP services. Existing wide area network (WAN) links may already be taxed to their limits supporting existing data applications. In many cases, these links may be using frame relay connections that might not be voice-friendly. Frame relay is very much a data services. The overall capacity in terms of bandwidth and processing power are vital factors to assess before implementing any new network service.
There is a pitfall to watch out for when implementing VoIP services. It's important to avoid the added expense associated with multiple upgrades. Given that for most organizations, network upgrades represent capital expenditures (CAPEX), no technical manager should be put in the awkward position of justifying multiple upgrades. A smart approach is to assess and understand the voice calling and quality requirements in conjunction with all existing data services. In short, invest the effort to do a complete, holistic assessment of the entire network services suite across the enterprise. This is also a good time to consider any planned new data services as well as the VoIP requirements. Rather than upgrading prematurely, focus early in the process on data gathering and information analysis. Taking a methodical approach will not only ensure successful pilot testing, it will help yield a more future-proof design with the ability to grow smoothly and support future, anticipated business needs.
Trunking capacity is always a consideration in voice networks. It's tied directly to carrying capacity. In the VoIP world, how and where gateways are connected to the outside voice network, or PSTN, are important performance and management factors. T1 circuits, whether primary rate (PRI) or standard channelized connections, can provide a standardized approach that traditional telecom providers support. This approach is simply building T1 trunks in the same way you connect an enterprise PBX to the PSTN today. In the traditional T1 environment, a T1 circuit can carry 24 voice calls.
IP trunking presents a different approach, using the full bandwidth of a network connection to the carrier. SIP trunking over IP is becoming very popular and quite common. SIP trunking is easy to implement in most current vendor offerings, and provides an easily quantifiable return on investment. In short, it's proving to be quick and easy to implement, and a powerful cost-savings approach. Given the codecs mentioned earlier, a 100Mb SIP trunk to the carrier, using a codec such as G.729a, can carry a high volume of voice traffic easily.
There's an old networking paradigm that has been used by many IT managers. When network performance seems sluggish, add bandwidth. This approach works in many cases, but it's usually a temporary fix. It's also an approach that, if followed over time, guarantees that network circuit costs will spiral ever upward. Bandwidth is an important characteristic in network performance, but it's not the single most important factor. Networks have a wide-ranging set of performance characteristics that determine how smoothly they operate.
There's a principle in network management, often related to security, which identifies three important network requirements—confidentiality, integrity, and availability (often called CIA). There are many different components and characteristics that play into achieving complete confidentiality, integrity, and availability in the enterprise network. These characteristics make up what I'll call the network performance envelope. The shape of this envelope gives a view into the "personality" of the network.
We've touched on the importance of knowledge about your network operating environment up to this point. This section will put together the pieces so that you can utilize a knowledge-based approach to ensuring success in deploying integrated VoIP services. The more you know about the network, the more accurate your planning will be, increasing your ability to meet call quality expectations for end users. Better data improves your ability to provide better total network performance.
The basic concept of confidentiality, integrity and availability doesn't offer enough granularity to fully assess the requirements for successfully delivering a converged VoIP service. To accomplish that, you'll need to look at a broader set of data inputs. You'll need specific, measurable data that you can assess to gain a complete understanding of both the existing network and the new requirements.
There are a nearly unlimited number of different data elements an organization might use or consider when defining the network performance envelope. This guide will use the factors shown in Figure 5.1 as an example and will explore some characteristics in more detail than others. These chosen factors provide a good representative sample of the types of characteristics enterprises might consider when working through this process.
Figure 5.1: Graphing performance characteristics.
This figure shows the basics of CIA in a granular form and brings some other network performance characteristics into the mix to ensure you can fully support converged VoIP services.
Reliability overlaps both the availability and the integrity of the network. Because it's so important, we'll treat reliability as a discrete evaluation factor here. Throughput also plays a dual role, providing both availability and integrity. We'll include three facets of throughput in the example:
We've added cost, which is unrelated to CIA in every way, but a business consideration that must be taken into account with every network change. Some enterprises might substitute a more comprehensive ROI model as part of this assessment. No matter how simple or complex your assessment mechanism, managers will have to answer for profitability and cost recovery of changes to network services.
You must asses the manageability of the network. Deploying new services that cannot be effectively managed is courting disaster and ensures failure. For many service delivery organizations, this correlates into a service level agreement (SLA), or contractual commitment.
Scalability of the network has been included here as a broad measure for future-proofing your design. Every business enterprise plans for business growth. Business growth inherently drives network growth as almost every business sector becomes more and more dependent on data networking resources. You're striving to future-proof the network for some period of time, typically 3 to 5 years. No network designer or engineer wants to implement a major new service such as VoIP on a network, only to discover a year later that the network was under-powered during a major project and now requires a complete redesign.
Integrity encompasses network reliability requirements. Security is included here as an aspect of integrity, encompassing three facets for consideration:
Availability may have many different meanings to many different people. To some enterprises, and most of the telecommunications carriers, it generally indicates a guaranteed uptime at the five nines, or 99.999 percent level. Rather than deeply explore network design issues such as high availability, redundancy, and business continuity, availability will be used here as another data point on a graph for analyzing the network performance envelope.
Figure 5.2 gives you the next step in the process. It presents the framework you can use for mapping each operational characteristic of the larger network performance envelope.
Network services and applications all have unique and different requirements. Email can be delayed several minutes without any measurable negative impact. Delays in Web browsing might be barely noticeable as the browser screen gradually loads information. Its use of Transmission Control Protocol (TCP) at a higher layer guarantees information delivery, although the overhead might add delay. Web traffic is non real-time traffic, so between a person and a system, performance impact may be negligible. Integrating VoIP into the often already overtaxed data network introduces a whole new set of requirements into the network performance envelope. As these new characteristics needed for new network services are identified, they can be mapped onto the graph. As you do so, notice that the personality of your network, the performance envelope, begins to take shape.
Figure 5.2: Identifying performance characteristic requirements.
Every network service and application in operation needs to be evaluated and factored into the performance envelope data set you're evaluating. Remember, you're identifying the requirements for successful service delivery—your success factors. You are planning to succeed. For existing services such as email and Web traffic, you might make some simple assumptions about acceptable performance. For mainframe applications, or those that were custom-developed in-house, it might be prudent to overlay the specific requirements to support each. The key is that each requirement be identified using some measurable value. The data points or elements used by any organization can be established as appropriate.
This model is demonstrated in a very simple format. Each network has different and unique requirements depending on the services provided and applications in use. If your network employs QoS mechanisms for different traffic types, don't forget to account for each of them when graphing out your network requirements. The more granular the data used, the more accurate the assessment of performance requirements becomes. As always, the more you know about the network, the better your decision will be.
Every performance characteristic you asses becomes a data point on the graph lines. In the next step, which Figure 5.3 shows, "connect the dots" to gain visual representation of the shape, or personality, of your network. You know what the network needs to look like to successfully deliver the existing service and applications in addition to VoIP and other new services needed.
Figure 5.3: Giving shape to the network performance envelope requirements.
For many organizations, now the easy part is completed. The next step in the process is to physically measure each data point on the graph. Some elements, such as cost and security, may be relative assessments rather than technical measurements. The key to success is to be methodical and thorough. .This step will quantify network performance capabilities today. As Figure 5.4 shows, when you map the real-world measurements from your network with the performance envelope you've established as a requirement, you have a gap analysis mechanism. This gap analysis can now help you focus directly on those areas of network performance that fail to meet your established requirements. This helps ensure a methodical approach to upgrading the network.
As part of the overall operational requirements of the new, converged network, security and performance management often mesh to become one overarching facet of design consideration. Achieving the delicate balance between service delivery and security requirements means that compromises and tradeoffs will be necessary. The right set of management tools will allow you to continually monitor network performance and security, assess risks, and measure performance around the clock.
Neither performance nor security can be effectively monitored as a single element within the network. The network is a large, almost organic environment. A systematic and holistic view of the health and welfare of the entire system requires vigilance and constant review. Beyond this methodical approach to assessing the network performance, requirements implementing repeatable, sustainable processes will help ensure consistent network performance that delivers both the quality and security necessary for widespread enterprise success. There are a variety of Network Management Systems (NMSs) in use in business networks today. These range from expensive commercial products to freeware and open source tools. Beyond network and security monitoring tools, there are a number of VoIP-specific solutions to aid in constant monitoring of the VoIP service. The best resource for identifying the tools that will work in your environment is generally the VoIP solution provider.
Figure 5.4: Gap analysis—Mapping existing performance against the requirements.
You can use this performance envelope graph to overlay the performance characteristics the network provides today with the requirements you've already documented. You know your requirements for successful service delivery and you know your network capabilities. Figure 5.4 demonstrates a common occurrence in many networks. The reality of network measurements and the requirements don't align in all areas. In some characteristics, the network provides better service than needed. This may mean that you're paying a premium price to a carrier unnecessarily. In other areas, there are gaps to address in order to meet service delivery needs.
Implementing a methodical approach such as the performance envelope when analyzing service requirements also helps in defining VOIP service delivery expectations. Clear expectations are a key factor to a successful implementation. This approach ensures that you know what is required.
Completing a thorough assessment of the existing network has two benefits. First, it provides an accurate and viable gap analysis to use in preparing the network for converged services. In today's frenzied network operations environment, it's sometimes too easy to assume that adding a switch here, a link there, and increasing the bandwidth somewhere else will meet service requirements. It's important to take the time to complete a thorough assessment and provide comprehensive gap analysis information to achieve success. As a side benefit, especially in large, distributed networks, some enterprises will identify areas in which networks have been overengineered in the past. This can lead to redesign and cost savings.
This performance envelope readiness assessment is an opportunity to re-evaluate the existing network and determine whether current needs are being met. Many corporate enterprise networks were not designed to be what they have become today. Networks often began as small islands of information—isolated workgroups or departmental LANs. As the network gradually grew, connections to other groups and organizations were added. In most organizations, new business applications were also added over time. Corporate networks have grown from a simple beginning into a complex and sophisticated mesh, weaving corporate operations together. Often they've done so without being revisited from a holistic, service delivery perspective. This performance envelope approach to network assessment can help lead to a network that delivers better performance at a lower cost.
Data, voice, and video services each place different demands on the network. They're different types of service. Each one is designed to provide a specific service. To support different types of service in the network, you need to be able to offer some consistent and predictable QoS that you can manage. This is often accomplished by creating a specific class of service for each type of traffic supported in the network. For some network designers, determining the number of service classes required can be a challenge. Although implementing QoS mechanisms isn't a convergence step that's required, many organizations find that VoIP drives the need in order to deliver the required performance characteristics.
QoS can become very complex. Implementing an overly complicated QoS scheme can lead to a network that can't be readily supported. Rather than add undue complexity, let's look at just some of the performance characteristics included in the performance envelope example.
As network convergence and emerging technologies gain momentum, be mindful of the danger that each new application brings. It's important to look toward the future and not create a new class of service for every new application, or network complexity can spin out of control. For manageability, most network engineers favor using only a few critical service classes.
VoIP and video collaboration represent real-time traffic. This is most often communications information flowing from person to person, rather than interaction with a server or system. These real-time services require quick delivery with quality assurances for delay, jitter, and loss.
IP networks use a best efforts approach to deliver all general traffic. This class of service is quite suitable for email, Web browsing, and most other normal network traffic. Best effort simply uses whatever network resources are available.
In some networks, management traffic may warrant a dedicated class of service all by itself. This approach is quite common in service provider networks. It provides a mechanism to ensure the network can always be managed, regardless of congestion problems that might arise.
All QoS mechanisms provide some form of traffic prioritization scheme. In the converged network, there are many different traffic types. Each may have different requirements and different prioritization needs. Similar traffic types need to be identified so that they can be handled the same way within the network. Most organizations choose to aggregate similar traffic types. This approach allows a company to take advantage of network routes that are optimized to provide the appropriate class of service.
In designing a VoIP service network, the focus is typically on call quality for the voice user, but it's important not to degrade pre-existing services when implementing VoIP. You need to recognize all the different traffic types in use. If mission-critical data applications aren't given the necessary resources through QoS prioritization, the applications might "starve" for lack of resources. Email and Web traffic will still need to be delivered, even if it's a lower priority. It's vital to maintain balance across all traffic types when managing an integrated data, voice, and video network.
Throughput is frequently measured in terms of bandwidth. When evaluating throughput requirements, due consideration must be given to traffic aggregation points. Don't overlook the congestion issues that can develop as a result of combining 10Megabit, 100Megabit, and Gigabit Ethernet connections on the network. This traffic, all flowing to a centralized aggregation point, may overwhelm a lower throughput link. This can aggravate network congestion and introduce service delivery problems.
Over-engineering or over-provisioning the IP network has been a common approach for many network engineers. Increasing bandwidth, by ordering higher capacity links, has been the most common technique. Adding bandwidth may alleviate short-term problems, but it's important to remember that IP uses all the available network resources. Bandwidth is one resource. Overengineering frequently proves to be a delaying tactic that simply stalls necessary redesign work. This approach can be more costly in the long run. Adding bandwidth still requires investment— upgrading equipment and increasing bandwidth of circuits. These can become very expensive approaches, and they don't solve the problem of design. IP data applications can quickly consume all available bandwidth, leaving the same congestion problem to be addressed.
Response time is one measure of network performance, most often measured using ping as a test tool. One important nuance in the converged voice and data network is the fact that voice is an end-to-end service between people. Ping can be an effective diagnostic tool, but the end-to-end nature of voice service, coupled with the fact that delay is cumulative, necessitate comprehensive management and testing to ensure service levels and call quality are maintained.
CPU utilization in network nodes offers a good indicator of the overall health of the network. In planning VoIP services, utilization might provide an indicator that network elements are overtaxed and can't effectively support VoIP. After VoIP services are live, ongoing monitoring of CPU utilization can provide a benchmark and timeline trend analysis to monitor the health of the network over the complete life cycle.
Ethernet is, at its roots, a shared media technology. LAN switching provided network technicians a means to segment traffic into smaller broadcast domains. This increased granularity is now often enhanced through the use of virtual LANs (VLANs). Like bandwidth, CPU utilization, and other factors, network segment utilization can be monitored as part of the day-to-day management operations to ensure adequate network performance to support the required services.
The integrity and reliability of the network encompasses a number of different technical facets. Each can be monitored, measured, and managed as a part of network operations. In the legacy IP network, these may have provided acceptable service and been left unchecked. In the converged service network, they warrant ongoing monitoring and proactive management
A common measure is error rate and data loss. When IP networks are used to transmit normal data—email, file transfer, Web browsing, and so on—some data loss is acceptable. The higher layer protocols, such as TCP provide a measure of quality assurance and request retransmission when needed. Other data types, mainframe data, as noted, may be very intolerant of packet loss.
Jitter describes the variations in delay. As IP networks route traffic over the network using the best path identified by routing protocols, it's possible that every packet in a stream might take a different route. A VoIP call could potentially traverse many network paths. Each route through the network may have different delay characteristics. Jitter typically isn't a concern for the normal IP data traffic c. VoIP is far more sensitive to jitter than email or Web browsing because it is a real-time service between people. High jitter can result in unintelligible conversations that sound "jerky." Users won't trust or use VoIP services if the call quality is unacceptable.
Delay exists in all IP networks. It exists in several forms. Routers use statistical multiplexing algorithms to process traffic. Assembling data into packets takes time. Checking the routing protocol to identify the best route through the network takes time. These miniscule delays add up and all impact the total end-to-end delay. Delay is cumulative.
No business can operate without maintaining vigilance in controlling cost or expense. In networking, you face not just CAPEX in hardware investment but also the operating expense (OPEX) of keeping the service up and running on a daily basis. Some businesses, notably service providers, focus on profitability and use ROI as a performance envelope measurement. Enterprise businesses may treat their IT and network operations internally as either a profit center or a cost center. For many, no profit is expected, but the costs associated with
implementing and managing the network still must be recovered. For most organizations, over a 3-to-5 year life cycle, network OPEX tends to be much greater than the initial CAPEX investment. It's important to always factor the appropriate cost analysis in both VoIP preplanning and the ongoing service management.
Availability, for many, is defined as reliability. It means that the network is fault tolerant and services don't degrade when problems do occur. To provide reliability in the PSTN, there are millions of circuit paths available through hundreds of central offices. In an enterprise IP network, redundant paths, alternative routes, load-balanced connections, and high-availability equipment may all need to be incorporated into the design to ensure resilience in the network.
Availability in traditional telephony has often been measured as the uptime percentage. In commercial networks, you've heard the term "five nines reliability" (or 99.999 percent uptime) used as the target availability measure. For enterprise operations, it's crucial to recognize that this number equates to roughly 5 minutes of downtime per year. Although the commercial telephone providers have widely met the five nines measure, there aren't many corporate data networks that can claim less than 5 minutes downtime in the past year. Introducing VoIP and other real-times services into the network raises the bar for IP network availability and drives designers to invest in robust, fault-tolerant design solutions.
When implementing VoIP solutions, network security is as great a concern as reliability and call quality. Corporate networks may include firewalls and multiple connection points. These security devices can add nodal processing delay that may impact VoIP services. The more complex a rule set in a firewall, the more latency it induces to the data flow. Remember that delay is cumulative and counts toward the 250ms maximum tolerable end-to-end delay.
VPN services are deployed in two typical fashions. Point-to-point VPN solutions may be used to connect remote offices over the Internet. Many companies also use VPN services allowing employees to connect to network resources while telecommuting or away from the office. Encryption algorithms consume processor power. A VPN device running DES or Triple DES encryption as a VPN endpoint will add further latency as packets are encrypted and decrypted. If the VPN endpoint is a firewall, this CPU load problem may be further compounded.
Security concerns reinforce the importance of assembling a comprehensive technical team for network assessment, readiness testing, and managing the operational environment. It's crucial that the telecommunications, IT, and network security teams collaborate to be successful.
The human resources aspect of supporting network services may be the mostly costly component. This is the biggest OPEX cost component. The more complex and difficult the network is to manage, the more resources required. As network designers and service providers, we must consider the ease of management we're incorporating into VoIP network design.
Scalability provides the best measure for future-proofing the network. Corporate networks change, grow, and evolve continually. They become very organic in nature as business needs change. The evolutions of the SOA and SaaS on the network are beginning to accelerate. The demands placed on the network will grow as applications, services, and networks become more tightly coupled with business processes. You must always consider the ease with which the network can scale to support new business applications and services.
The key to successfully managing the converged network begins during the planning process long before implementation. Conducting a needs analysis will help define the business needs in measurable terms. Completing a methodical network readiness assessment will further develop a holistic vision of what the network must support in order to successfully deliver integrated services. Don't sidestep these important steps early in the VoIP deployment process. They will not only ensure that the converged network fulfills your business needs but also help formulate the day-to-day management methods required to support the enterprise integrated services.
Perhaps a key differentiator in using this performance envelope approach to network performance analysis is that it offers a tool to support all services in the network. VoIP is important, but unified communications is only effective when it supports the larger unified application environment that supports the complete business of the enterprise. It offers a holistic view of not just data voice and video but also enterprise-specific applications and services. Sales force automation, inventory management, customer relationship management, and Web-based ecommerce are examples of vital application services that the enterprise needs in order to thrive. Although the software-oriented architecture mindset will more tightly couple these applications with services such as voice and data, using the performance envelope as a holistic approach is an integral part of managing a converged network, encompassing the full spectrum of data, voice, video, applications,
This chapter only briefly touched on the idea of using outside resources. It's worthwhile, in summary, to review how outside integrators and consultants can work for you. They can help look out for your best interests. There are many VoIP consulting firms that specialize in converged technologies. Their reputation is built on how well they take care of client needs. They build business through word of mouth referral and will be your advocate for success. They can ensure that you are being well served by your vendors.
Your equipment vendors and service providers will gladly offer consulting services, and their sales engineers will graciously come in and help you design a solution. It may be pertinent to recognize that vendor staff may be specialists in their company-specific products. They may do a very good job based on their own experience, but they might not have comprehensive training on the workings of competitive solutions. Use your trusted and incumbent vendor partners, but don't give them free rein to design your network. You need to remain in control of the project overall, because you'll have to manage it in day-to-day operations.


An expanded version of this tutorial is available for MixEffect University members: https://labs.mixeffect.app/lesson/directly-connecting-mixeffect-to-an-atem-using-ethernet/
Learn how to directly connect MixEffect to your ATEM switcher via Ethernet. You may need to do this if you are out in the field for a video production and there’s no network for each device to plug into.
In this video, you’ll learn how to:
- Connect an Ethernet cable between your iPhone/iPad and your ATEM switcher.
- Configure a static IP address for each device.
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The CSIR’s research in the field of Optronic Sensor Systems (OSS) focuses on new and novel electro-optic sensors, and the modelling, simulation, engineering, testing, evaluation and development of advanced electro-optical sensor systems for day, night and multispectral surveillance. Researchers also evaluate and design countermeasure and electronic warfare techniques in the visual and infrared wavelengths. To underpin this work, the CSIR has a suite of specialist facilities, including a test and evaluation laboratory, flight motion simulator, ship motion simulator and missile exploitation laboratory.
Research areas include self-protection, advanced optical systems, surveillance applications and information application especially the management of large amounts of image data.


It is a review of Samsung Exynos 990 in which you will learn its technical specifications and benchmark.
Its standard number of cores is 8. The base clock speed of Samsung Exynos 990 is 2.00 GHz whilst its clock speed in turbo boost is 2.73 GHz.
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Besides being used to access websites, an IP address is also utilized in many other ways. It is an important component of a computer network and helps in the identification of a device. The IP address is assigned to a device and changes whenever the device changes its location or network connection.
The most obvious function of an IP address is to provide a unique identifier for every internet connected device. This enables computers, tablets, phones, and other devices to communicate with each other. It is also a requirement for any webpage to be opened. In addition to identifying a device, an IP address can be utilized to discover the name of the owner of a certain website. In addition to being a tool for identification, an IP address is also useful for tracking a user’s location and to identify the country the user is in.
There are two types of IP addresses. The first type, known as a public IP, is usually used only for surfing the web. The second type, called a private IP, is used to allow devices to connect to a particular network. A private IP is usually used in a home or local area network.
The Internet Corporation for Assigned Names and Numbers, or IANA, is a non-profit organization that administers the IP addresses of the world’s public internet. IANA, along with other entities such as ICANN, is responsible for the maintenance of the security and usability of the internet. Besides ensuring that the internet works properly, ICANN is also a non-profit organization that is funded through fees paid by domain name registrars.
The smallest and most elusive of all IP addresses is the IPv6. This IP address check type of address is the most sophisticated and is equipped with many features that aren’t found in the more traditional IPv4. It has a Maximum Transmission Unit (MTU) of 1280 bytes and a checksum field. This address also has a few other features, including packet flow identification and renumbering. However, there is a downside to this type of address. Because an IPv6 address has a relatively small size, it can be difficult to use.
The most common ways to get a device’s IP address is to hack the device itself or to visit a site that allows the user to view their IP. Other methods include obtaining the address through malware or clicking on malicious links. There are also apps that can be used to find an IP.
It is not uncommon for cybercriminals to steal an IP’s identity by placing a bug in an HTML email or by using the social media to pinpoint an individual’s location. They can also download content that is illegal or terrorism-related. This can be done through the use of instant messaging applications or by hacking a home network.
A good way to discover what you can do with your IP is to do a reverse IP lookup. This type of service is available for free and will allow you to determine the name of the owner of an IP, the city in which it is located, and the contact details of the owner.


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Sometimes I feel like God’s relationship with me is like my relationship with my cell phone.
There are times I’ll be riding in the car through an area packed with businesses, and I want to check my email. My cell phone tires, to no avail, to connect to the internet using an “unlocked” and weak WiFi network that it might have found 30 feet away. My cell phone does this without my approval, supposedly to save me data, or maybe it’s just being too lazy to connect to the cellular network.
I wish my cell phone would trust me. For I have paid, sometimes dearly, for its reliable cellular network and safe WiFi networks where it can find rest for its weary connection-finder. But my cell phone feels it can do this and other things on its own, such as dictating and interpreting my words, understanding where I want to put the cursor, and when I want the screen horizontal.
If only my cell phone would learn to trust me and the internet connections I have paid to provide it, as well as the gentle commands I give it, it would serve me well and it would live a long and prosperous life. For I will not forsake a cell phone before it is well-utilized.
I am actually a caring cell phone user. I don’t exhaust the gigabytes with photos, videos and applications. I make efforts to keep dust from getting in ports and I buy cases when necessary. I have access to a plethora of cell phone doctors. I haven’t and never will use it for more than what it’s built for.
So, if I could say something to my cell phone, it would be: “Please don’t exhaust yourself. You’re a wonderful creation and I yearn for your service to me. But you must learn to trust how I’ve made you, my provisions for you, and my commands to you.”
Anything you want to add?


|会場||東京農工大学 小金井キャンパス 13号館 4階 講義室 L1341|
|講演者・講演タイトル||"Multi-antenna Interference Management for Coded Caching"
Dr. Antti Tölli
Associate Professor, Centre for Wireless Communications, University of Oulu (Finland)
Antti Tölli (M'08, SM'14) received the Dr.Sc. (Tech.) degree in electrical engineering from the University of Oulu, Oulu, Finland, in 2008. Before joining the Centre for Wireless Communications (CWC) at the University of Oulu, he worked for 5 years with Nokia Networks as a Research Engineer and Project Manager both in Finland and Spain. In May 2014, he was granted a five year (2014-2019) Academy Research Fellow post by the Academy of Finland. Currently, he also holds an Associate Professor position with the University of Oulu. During the academic year 2015-2016, he visited at EURECOM, Sophia Antipolis, France, while from August 2018 till June 2019 he is visiting University of California - Santa Barbara, USA. He has authored numerous papers in peer-reviewed international journals and conferences and several patents all in the area of signal processing and wireless communications. His research interests include radio resource management and transceiver design for broadband wireless communications with a special emphasis on distributed interference management in heterogeneous wireless networks. He is currently serving as an Associate Editor for IEEE Transactions on Signal Processing.
In Coded Caching (CC), instead of simply replicating high-popularity contents near-or-at end-users, the network should spread different contents at different caches such that common coded messages broadcast during the network high-peak hours to different users with different demands would benefit all the users simultaneously. In this talk, a multi-antenna broadcast channel scenario is considered where a base station delivers contents to cache-enabled user terminals. A joint design of CC and multigroup multicast beamforming is proposed to benefit from spatial multiplexing gain, improved interference management and the global CC gain, simultaneously. The developed general content delivery strategies utilize the multiantenna multicasting opportunities provided by the CC technique while optimally balancing the detrimental impact of both noise and inter-stream interference from coded messages transmitted in parallel. Moreover, reduced complexity alternatives, with a small loss in performance, are introduced by controlling the size of the subset of users served during a given time interval, and the overlap among the multicast messages transmitted in parallel, indicated by parameters α and β, respectively. Degrees of freedom (DoF) analysis is provided showing that the DoF only depends on α while it is independent of β. The proposed schemes are shown to provide the same degrees-of-freedom at high SNR as the state-of-art methods and, in general, to perform significantly better, especially in the finite SNR regime, than several baseline schemes.
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COPYRIGHT 2020 ©TOKYO UNIVERSITY OF AGRICULTURE AND TECHNOLOGY., ALL RIGHTS RESERVED.


If you look at the Release Notes for 8. Once the IOS for the 1800/2800/3800 stabilizes, cisco ap 1800 manual Cisco is going to announce the RMA for the 1700/2700/3700. The Cisco Aironet 1800i access point is referred to as the access point, or abbreviated as AP in this document.
capwap ap ip address capwap ap ip default-gateway Also you can specify controller ip address: capwap ap controller ip address If ap is already joined to controller then login to WLC via web browser, click wireless tab, select AP, and in part of cisco ap 1800 manual cisco ap 1800 manual "IP Config" you can check "static IP" and set your desired IP address. com 1-14 Documentation DVD cisco ap 1800 manual 1-15 Ordering Documentation 1-15. The Wireless Service Assurance platform has three components, namely, Wireless Performance Analytics, Real-time Client Troubleshooting, and Proactive Health Assessment. Your AP LED indicator is blinking amber.
The LED indicators of Cisco 1830 Access Point mean several status. I have download the app suggested from manual of the device but i cannot view this ssid. It means that AP priming to a new regulatory domain by Neighbor Discovery Protocol (NDP), in progress.
Manuals; Brands; Cisco Manuals; Wireless Access Point; Aironet 1800i; Cisco Aironet 1800i Manuals Manuals and User Guides for Cisco Aironet 1800i. 800 Series Router Cabling and Setup. An attacker could exploit cisco ap 1800 manual this vulnerability by sending a crafted IP packet to an affected device. Overview The Cisco 1800 series integrated services routers includes the Cisco 1841 router, which is a data-only router. If you want to contribute, please mail your pdfs to Cisco 1800 users, buyers and readers can check Cisco 1800 manual here, also Cisco 1800 router datasheet download is allowed. We have 1 Cisco Aironet 1800i manual available for free PDF download: Getting Started Manual. The wireless Cisco Aironet 1800s Active Sensor is a part of Cisco&39;s Wireless Service Assurance solution. Cisco 1841 routers support WAN interface cards (WIC), voice/WAN interface cards (VWICs) cisco ap 1800 manual in data mode only, single-wide high-speed WAN interface cards (HWICs), and advanced integration module (AIMs).
CISCO MOBILITY EXPRESS 1800 series AP. GETTING STARTED GUIDE Cisco Aironet cisco ap 1800 manual 1800s Network Sensor First Published: Ap 1 About this Guide About the Network Sensor Safety Instructions Unpacking the 1800s Network Sensor Ports and Connectors on the cisco ap 1800 manual 1800s Preparing the AP for Installation Installation Overview Performing a Pre-Installation Configuration Mounting and Powering the Network Sensor Configuring cisco ap 1800 manual the Network Sensor for. 概要 AP-OS AP cisco ap 1800 manual (AP18xx/28xx/38xx) を利用する際に集中管理型の CAPWAP イメージから Mobility Express イメージへ変換できないという問題が発生する場合があります。このドキュメントではその対処方法を紹介します。 実機での検証はバージョン 8. The vulnerability is due cisco ap 1800 manual to improper input validation of IP packet headers. The 3600 series access point is a controller-based (Unified) product and supports: • Simultaneous dual-band (2.
11ax (Wi-Fi 6) Access Points ; Install and Upgrade. Cisco Wireless Solutions Software Compatibility Matrix ; Feature Matrix for Wave. 2 About the Access Point The Cisco Aironet 1800i access point is an 802.
We have 9 Cisco cisco ap 1800 manual 1800 Series manuals available for free PDF download: Manual, Hardware Installation Manual, Regulatory Compliance And Safety Information Manual, Configuring, Quick Start Manual,. View online or download Cisco Aironet 1800s Getting Started Manual. 4 GHz/5 GHz) radios † Integrated antennas on the 3602I access point model (AIR-CAP3602I-x-K9) † External antennas for rugged 3602E access point model (AIR-CAP3602E-x-K9) Note The ‘x’ in the model numbers represents the.
Heathkit AP 1800 Schematic. A great leap forward from Cisco’s 1700 Series routers, the 1841 router increased performance by a factor of five. It is a part of Cisco’s popular 1800 series of the company’s selection of integrated service routers. 800 Series network router pdf manual download. User manuals, Cisco cisco ap 1800 manual Wireless Access Point Operating guides and Service manuals. Cisco 1800 Series Routers Models Comparison Cisco 1800 Series Routers Hardware Installation Guide (Modular) Cisco 1841. We have created TFTP point and tried updated firmware.
This is not very good for customers, it is not clear why not to leave this function. Learn what you need; View At-a-Glance. The cisco ap 1800 manual access point can be cisco mounted on a wall and supports 2x2:2 SS MU-MIMO applications. We also tried HTTP. 3 you&39;ll see the list as long Chinese take-away menu about bugs cisco ap 1800 manual for the 1800/2800/3800. which is the &39;go to&39; AP to deploy? The Cisco Aironet 1800i access point is referred to as the access point, or abbreviated as AP in this document. 2MR1 を使って実施しています。 TFTP サーバが別途必要です.
The Cisco Aironet 1800i access point is referred to as the access cisco ap 1800 manual point, or abbreviated asAP in this document. 0 firmware version and we tried to update to 8. The vulnerability is due to insufficient validation of PnP server responses. This module describes how to configure radio settings on a Cisco 800, 1800, 2800, or 3800 series integrated services router, hereafter referred to as an access cisco ap 1800 manual point (AP). This is the 8 pages manual for Heathkit AP 1800 Schematic. View online or download Rts ROAMEO cisco ap 1800 manual AP-1800 User Manual.
Re: AP-1800 Sensor AP mode I also cisco ap 1800 manual discovered that APs 16 have the same issue in 8. Conditions: Profiling based cisco ap 1800 manual on CDP attribute cdpCachePlatform, gathered by SNMP probe or IOS device sensor via RADIUS probe. A vulnerability in the IP ingress packet handler of Cisco Aironet 1800 Series Access Point devices could allow an unauthenticated, remote attacker to cause a complete denial of service (DoS) condition.
We have a Cisco Aironet 1830i access point. see product cisco air-ap1800s-e-k9 - cisco aironet 1800s series - network sensor, find price of cisco aironet 1800s series - network sensor, cisco air-ap1800s-e-k9 - cisco aironet 1800s series - network sensor. We help cisco ap 1800 manual companies of all sizes transform how people connect, communicate, and collaborate. View and Download Cisco 800 Series quick start manual online. With their intelligence, security, and superior performance, the new Cisco Catalyst 9100 Access Points are the beginning of your Wi-Fi 6 (802. Mounting the Sensor using AIR-AP-BRACKET-NS The Cisco Aironet 1800s wireless sensor can be mounted, in a vertical orientation, on a wall or desk, to a height of 3 feet, using the wall-mount bracket AIR-AP-BRACKET-NS. How to configure standalone Cisco access point from GUI and configure WPA2 Cisco is cisco ap 1800 manual the worldwide leader in IT, networking, and cybersecurity solutions. To mount the sensor, follow these instructions: Step 1 Identify the location for mounting the sensor.
11 a/b/g/n/ac (Wave 2) access point, with internal antennas only. The PnP feature is only active while. A vulnerability in the Plug-and-Play (PnP) subsystem of the Cisco Aironet 1800, cisco ap 1800 manual 2800, and 3800 Series Access Points running a Lightweight Access Point (AP) or Mobility Express image could allow an unauthenticated, adjacent attacker to execute arbitrary code with root privileges. Cisco Manuals; Network Router; 1800 Series; Cisco 1800 Series Manuals Manuals and User Guides for Cisco 1800 Series. Every time we try preimage download starts and attempts several times to complete but never does.
Download 454 Cisco Wireless Access Point PDF manuals. And we&39;re now shipping our newest Wi-Fi 6 cisco ap 1800 manual access point, cisco the Cisco Catalyst 9105 Access cisco ap 1800 manual Point. About the Access Point The Cisco Aironet 1800i access point is an 802. Install and Upgrade Guides; Cisco Aironet 1800i Access Point. Finding Feature Information in This Module Your Cisco IOS software release may not support all of the features cisco ap 1800 manual documented in this module.
02 MAY/ ROAMEO Family Wireless Intercom System •TR-1800 •AP-1800. The 1800 series, in turn, is but one of many series in the company’s larger portfolio of integrated service routers. Symptom: ISE Profiler is classifying Cisco Wave 2 APs, such as 1800/2800/3800, as Cisco-Switch instead of respective Cisco-AP-Aironet-1800 and so on. The Cisco Aironet 1800 family of access points, designed for small and midsize networks, delivers the performance you need for the ongoing proliferation of Wave 2 as well as support for older Wi-Fi devices. This guide also provides mounting instructions and limited troubleshooting procedures.
Cisco 1800 Series Integrated Services Routers (Fixed) Software Configuration Guide OLCONTENTS Preface 11 Audience 1-11 Organization 1-12 cisco ap 1800 manual Conventions 1-13 Notes, Cautions, and Timesavers 1-13 Command Conventions 1-13 Related Documents 1-14 Obtaining Documentation 1-14 Cisco. Mounting the Sensor Using AIR-AP-BRACKET-NS. Read or download the pdf for free.
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Real time embedded software & algorithm development
FFEI have in house expertise to develop image/colour processing and for the development of complex instrument control systems. This includes embodiment across many platforms including digital signal processors and fully integrated computing systems.
Real time embedded software
To provide effective machine and high speed data controls the software systems are based on real time operating system architectures which are integrated or embedded into our products. This includes an ability to both develop driver and kernel level software.
Digital images have to be captured, generated and manipulated. This requires a high level of competence in digital electronic design and implementation using high Level language descriptions targeted at Complex Programmable Logic Devices (CPLD) and industry leading Field Programmable Gate Arrays (FPGA).
In addition we also address all aspects of electronic analogue and digital system design from concept through to PCB production and integration.


A few examples include Ethernet. TCP treats the data as a stream of bytes. The link layer is used to move packets between the Internet layer interfaces of two different hosts on the same link.
Others are protocols for doing specific tasks, e. This is accomplished with a hierarchical IP addressing system. This four tuple uniquely identifies a connection. Internet layer The internet layer exchanges datagrams across network boundaries.
The protocol was implemented as the Transmission Control Program, first published in Protocol versions[ edit ] Versions of the Modbus protocol exist for serial port and for Ethernet and other protocols that support the Internet protocol suite.
Alternatively, the packet might get queued up until a thread becomes available. The web server receives that request and sends the web page information back to the web client. The Network Access layer manages all the services and functions necessary to prepare the data for the physical network.
Lets take a very simple example to make the concept more clear. The Internet protocol suite and the layered protocol stack design were in use before the OSI model was established.
Join request will then be sent to coordinator. A half dozen routers might be configured to connect several dozen departmental LANs in several buildings. It becomes the responsibility of the sender to retry the data a few seconds later and to persist until it finally gets through.
User protocols are used for actual user applications. Such a connection, or virtual link, may be established with a transport protocol or even an application scope protocol that serves as a tunnel in the link layer of the protocol stack.
InRobert E. This is also the layer where packets may be selected to be sent over a virtual private network or other networking tunnel.
Key architectural principles[ edit ] An early architectural document, RFCemphasizes architectural principles over layering. So this way TCP makes sure that no packet gets lost.
This is the basic task of sending packets of data datagrams from source to destination by forwarding them to the next network router closer to the final destination.If you want o understand the INTERNET or any NETWORK you must know what IP cheri197.com’s why today i have decided to write a short note on IP here.
What is IP? IP is communication protocol of TCP/IP suit which is used within a network or outside a network or between two or more networks. Short note on TCP/IP The full name of TCP/IP is Transmission Control Protocol/Internet Protocol. And as name suggest it is responsible for.
The base D-Bus protocol is a one-to-one (peer-to-peer or client-server) protocol, specified in the section called “Message Protocol”.That is, it is a system for one application to talk to a single other application. Serial Line Internet Protocol (SLIP) (Page 1 of 3) The need for a data link layer protocol to let IP operate over serial links was identified very early on in the development of TCP/IP.
TCP/IP was and is the crown jewel of the US engineering acumen, the technology that changed the civilization as we know it in less then 50 years.
x and Beyond! IEEE x has been leveraged for a long time for authentication purposes. Up until this point, little has been done to help researchers expose vulnerabilities within the systems that implement the protocol.Download


The largest SIM card manufacturer will be acquired
French aerospace company Thales SA and Gemalto SA from the Netherlands jointly announced that two sides have reached an agreement to buy for 51 euros a share for 4.76 billion euros ($5.6 billion)。
Gemalto,founded in June 2006,is a combination of Axalto and Gemplus,the world's two largest smart card manufacturers.It focuses on digital security.Gemalto mainly provides SIM card,EMC bank card,e-passport,NFC and other portable devices that can store applications and information,also with the premise of security.
As early as the 1990s,Gemalto's predecessor began to cultivate in China's telecom market and provide services for China's telecom operators.After the merger in 2006, the newly established Gemalto,based on the traditional SIM card business, turned its products into software and solutions,in which OTA technology was adopted.
In September 2009,Gemalto announced to provide UICC(universal identity encryption computer)and R-UIM (mobile user identification card) for China Mobile,China Telecom and China Unicom,become the first supplier approved to support different 3G wireless technologies such as cdma-evdo,td-scsma and WCDMA.
In the 4G era, Gemalto is still providing SIM cards for three major operators in China.With the advent of bike sharing, Gemalto entered into a partnership with Moby bike on June 30,2017.Its cinterionm2m module and machine identification module are embedded in the Moby bicycle smart car lock, so that position data can be transmitted and the car lock can be controlled remotely.
In addition, NFC is also a major area of Gemalto.In February 2013,Gemalto reached a cooperation with China UnionPay,announcing to build an NFC mobile payment ecological environment.This cooperation covers all key aspects of project deployment,including interoperability,security of TSM (trusted service management) platform, guidelines for project implementation, and technical standards and specifications.
Data security issues
For Gemalto, data security is always the core issue.
In February 2015, according to relevant documents and data provided by Snowden to the intercept in 2010,U.S.and British spies have hacked into Gemalto's internal computer network to steal the key used to protect the privacy of global mobile communications,these stolen encryption keys allow intelligence agencies in the U.S.and Britain to monitor the mobile communications of SIM card users without the permission of communication companies.
After the event, Gemalto acknowledged that the company's network may be compromised, but the impact of the intrusion activity is limited,Because it was the outdated 2G SIM card that was attacked. Since then,Gemalto has been more careful about safety.
Thales Sa's acquisition of Gemalto focuses on the security products, which can help businesses and governments prevent data theft and identity theft.It is understood that in addition to Thales, another company named Atos SE is also planning to acquire Gemalto.Atos SE offer of EUR 46 per share.
For Thales,the acquisition of Gemalto marks a significant strategic extension of the former in data security construction.According to Patrick Caine, CEO of Thales, the merger aims to become the world's leading provider of information security solutions.
At present, Thales's acquisition of Gemalto has been unanimously supported by Gemalto's board of directors, but the final completion of the transaction requires regulatory approval, which is expected to take place in the second half of 2018.


"A powerful, free mobile phone manager, to do all kinds of task from your PC if you have a data cable with your phone."
As you may or may not already know, FloAts Mobile Agent is a powerful, free mobile phone manager/administrator that lets you do all types of tasks from your PC if you have a data cable with your phone. Its functions cover from monitoring to remote control and organization, etc.
It is very easy to use. FloAts Mobile Agent automatically detects your phone (when connected via the cable) letting you manage it via Windows. You can read and send SMS as if it were Outlook, administer your contacts agenda quickly and easily, or upload data and games to your phone.
Once connected you have access to a multitude of details of your phone: battery level, temperature, cover, contacts, calls, files, etc. Any data stored in your device, whether in the memory or SIM card.


|System||An autonomous buoy powered by Lithium primary batteries and lead-acid rechargeables replenished through a solar array.|
|Location||Test deployment in the Elson Lagoon at Barrow, Alaska. (71.3406° N, 156.545° W)|
The O-Buoy project is a collaborative effort to develop an autonomous buoy to measure atmospheric chemistry. The O-Buoy has evolved considerably from the first concept (left below) to the buoy that was installed in February 2009 (right below). The buoy has several components including the main buoy housing, a floatation collar, and a mast where instruments are mounted. The power system was a joint development by Purdue University and the U.S. Army Cold Regions Research and Enginerring Laboratory.
A Web-based interface to track instrument data and system performance: http://transport.sri.com/obuoy/monitor
The project Website with detailed information about the instrumentation and power system. The site has many photos of the buoy structure and deployment.
International has developed a communications controller, utilizing a
low power single board computer running Linux, and an Iridium A3LA-SA
transceiver. This system provides scheduled data transfers from the
buoy along with updates uploaded to the instruments.


CISCO 1841 ROUTER CONFIGURATION GUIDE EBOOK DOWNLOAD
13 Jul Step 3: Now let’s configure up both the WAN and LAN interfaces on the router like so: MICHAEL#conf t. Enter configuration commands. 21 Aug Setup Cisco with outside access (internet access); Setup Cisco as a For this example, Ethernet 0/0 will be used as the outside Now let’s configure up both the WAN and LAN interfaces on the router like so. For Cisco Router, you can follow these steps following. 1. Enter configuration commands, one per line. Router(config)#hostname MICHAEL
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Nope, unless you are trying to use that thing as a switch. Mon Mar 07, They’re being plugged into Catalyst switches whose exact model number I can’t remember right now.
Jul 13, Posts: Yeah, I’m a big Cisco newbie right now, but I’m trying to get to where I can stand on my own a little better and quit bringing in the high-dollar outside guys who used to do all of our Cisco configurations. Jan 26, Posts: Cisco 1841 router configuration guide on what you are trying to do. You have two fast ethernet interfaces on the router, how about one on the maintenance vlan and one on the regular network?
The two onboard ports cannot be used as switchport ports.
Vlan interfaces on a router like that when supported appear as sub-interfaces to the physical interface. I’ll try doing some research on Cisco’s site on dot1q stuff.
Cisco VLANs – Ars Technica OpenForum
Compiled Wed Oct They can only be used as routed ports. So I believe my problem is most likely that I need another FastEthernet port on a modular card, instead of trying to use one of the 2 interfaces built rouer the router already, if I understand correctly.
Cisco 1841 router configuration guide the end, I’ll have a North, South, and Central site, all with devices plugged into certain ports of the Catalyst s and routeer between each other with the Cisco routers. Originally posted by erratick: Sat Mar 05, 6: What type of switch are you plugging the routers into?
Sat Mar 05, Originally posted by monkyman: Can anybody give me a step by step guide for setting up multiple VLANs on a Cisco modular router?
*How To* Configure and get started with Cisco |
Here’s a link for configuratkon on the general config that you want to do: And for dot1q search cisco. Am I way off base somehow? Posting Guidelines Contact Moderators. I can’t seem to find the equivalent of this guide: Sun Mar cisco 1841 router configuration guide, Actually it depends on the hardware.
Additionally, you icsco have to actually perform a “int vlan x” before the vlan interface will be created. The onboard ports however cannot be made into switchports and are only supported as routed ports, which is where I think the OP is running into a problem.
It is different than configuring a layer 3 switch router interfaces which show up as int vlan x SpacemanSpiff Ars Centurion Tribus: Originally posted by LargeSock: Mon Mar 07, 1: System Bootstrap, Version Sun Mar 06, 2: Wed Mar cisco 1841 router configuration guide,
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What is Flightful Free messaging plan?
A: Flightful’s basic service provides free messaging on Signal, WhatsApp, Messenger, iMessage, and Google Messages. It allows text-only messages and emoticons. Images/videos, GIFs or other media, and other types of attachments are blocked. To access more apps and services, subscribe to Premium plan. To surf the web, upgrade to full internet access via in-app purchase.
How can I send and receive messages on Flightful?
A: Switch your phone to airplane mode. Connect to a Flightful compatible network through Wi-Fi and turn Flightful ON. You can start using messaging apps normally.
Why can’t I send photos and videos? / My photos and videos are not delivered.
A: Flightful’s Free plan only allows text-based messaging as a basic service. To send and receive all media, and other types of attachments, subscribe to Premium plan. To surf the web, upgrade to full internet access via in-app purchase.
Can I send and receive email while connected to Flightful?
A: Flightful’s Premium plan allows you to send and receive email messages on Gmail, Outlook, Apple Mail, Yahoo, and AOL. See the updated list of apps and services supported in the Premium plan here.
Why can’t I use Facebook and other social media apps?
A: Flightful’s basic service is only text-based messaging on WhatsApp, Facebook Messenger, iMessage, and Google Messages. To use Facebook and other social media apps, you need to subscribe to full internet access available via in-app purchase. See the full list of supported apps in Free and Premium plans here.
Where can I use Flightful?
A: Flightful can be used on inflight Wi-Fi networks onboard aircraft equipped with Telekom Flynet. Note that not 100% of flights are connected yet, so service availability can vary depending on the aircraft used by the airline.
Which airlines are supported?
A: Flightful can be used on in-flight Wi-Fi networks onboard of the following supported airlines (Note that not 100% of flights are connected yet, so service availability can vary depending on aircraft used by the airline.): Aer Lingus, Air Europa, Austrian Airlines, Lufthansa, TAP Air Portugal, United Airlines, & United Express.
How do I know if I am connected?
A: You can test your Flightful connectivity in-flight through the Flightful Bot available in WhatsApp and Messenger. Add Flightful Bot to your WhatsApp contacts and the Flightful Test Bot to your Facebook Messenger.
Type and send help and you will receive a reply of available commands to test your connectivity. This means you are successfully connected! If you do not receive a reply, check if the:
1. Wi-Fi is ON
2. Flightful compatible network is present and that you are connected to it
3. VPN is ON
4. Status on the home screen is online
I cannot see a Flightful compatible network.
A: This means that your flight is not supported. All supported networks are annotated in device Wi-Fi settings as “Flightful compatible”.
There is no connection or my connection keeps dropping.
A: In most airlines, the internet connection is provided only if the aircraft is at cruising altitude (above 10000 ft.). Due to technical limitations of satellite technology, there can be interruptions in connectivity during the flight (ie. high latitudes, aircraft turns etc.).
Can I purchase full internet access?
A: Yes. Flightful offers full internet access via in-app purchase on iOS. Internet access on Android will be available in Q2 2021. However, if you choose to purchase internet access from the airline or any other provider, switch Flightful OFF and proceed with your transaction.
Can I use my downloaded content while using Flightful?
A: Yes, all pre-downloaded content, ie. Netflix series, YouTube videos, Spotify playlists, podcasts etc. can be used normally while using Flightful.
My airline provides me free internet access. Why do I need Flightful?
A: If you have access to the full internet from your airline, then you don’t need Flightful on top of that. Just log in to the internet following the airline provided instructions.
Should I switch Flightful OFF when my flight is landing?
A: Flightful works only when the aircraft is connected. Airline connectivity is usually turned off during take-off and landing. Unless the airline provides gate to gate internet access, Flightful will not work. We suggest turning Flightful OFF when you’re not using it, just like when you log off your computer at the end of your day.
How can I use my flight’s onboard entertainment?
A: Usually airlines provide onboard entertainment using their own apps. Onboard entertainment works in most cases in parallel with Flightful. However, we suggest turning Flightful OFF if any problems occur with airline provided entertainment.
What phone models are supported?
All iPhone 6s and newer running iOS 13.
All Android devices running Android 6 and newer.
How do I set up Flightful?
A: Download and install the app from the AppStore/Google Play Store and register with your mobile number.
I did not receive my code or the code I received is not working.
A: After 15 seconds you can click “New code” and a new one will be sent to you. If code was not received due to an error in typing your mobile number, click “Start again”.
Why do I need to install the VPN profile to my device?
A: VPN technology is used to establish a secure connection which is essential for Flightful to work.
How do I turn off low power or battery saving mode?
A: Low power mode or battery saving mode can be turned off from your phone settings under Battery settings.
How can I register for Flightful in the middle of the flight?
A: Flightful registration is only possible if you have cellular coverage and your device can receive an SMS.
I deleted Flightful or I changed my device and would like to use it again. How can I set it up?
A: Just download Flightful app from Apple AppStore/Google Play Store and reinstall it to your device.
Is Flightful safe to use?
A: Yes. Flightful uses VPN technology to encrypt all traffic between your device and ground station. Flightful is not collecting any user data nor sharing it with 3rd party providers.
How is my data being used?
A: The only user data that Flightful has is your mobile number. This is used only for verification. No personalised data about usage of the service is collected by Flightful.


The launch date has slipped. Possible launch dates now are Nov 1 or 2.
The beacon will transmit every 15 seconds on both UHF and VHF frequencies.
- Spacecraft Name: HiakaSat
- Target Launch Date: Nov 1 or 2
- Orbit: 415x490km, 94.7 degrees, Period of 93.6 minutes
- UHF Beacon: 437.2705MHz (437.26-437.28 MHz), 9600bps GFSK 2W
- VHF Beacon: 145.9805MHz (145.97-145.99 MHz), 9600bps GFSK 1W
- Baseline Receive Equipment:
- Ham Radio IC-9100 or IC-7000, ‘Data 2’ Port 9600bps FSK Output
- MixW Software TNC in packet mode w/ VHF 9600bps defaults and TNC emulation, or Kantronics TNC using 9600bps FSK port