Black Box Explains...10-GbE, CAT6A, and ANEXT.
The IEEE released the 802.3an 10GBASE-T standard in June 2006. This standard specifies 10-Gbps data transmission
over four-pair copper cabling. 10-Gigabit Ethernet (10-GbE) transmission includes up to 37 meters of... more/see it nowCAT6 cable (with installation mitigation techniques), 100 meters of Augmented Category 6 (CAT6A) UTP or F/UTP cable or 100 meters
of S/FTP CAT7/Class F cable.
CAT6A is the ANSI/TIA 10-Gigabit Ethernet (10-GbE) over copper standard. Its requirements are covered in ANSI/TIA-568-C.2 (Balanced Twisted-Pair Communications Cabling and Components Standard) published in August 2009. It defines 10-Gigabit data transmission over a 4-connector twisted-pair CAT6A copper cable for a distance of 100 meters.
Category 6A cabling is designed to support next-generation applications, including the transfer of large amounts of data at high speeds, up to 10 Gbps. CAT6A extends electrical specifications to 500 MHz from 250 MHz for CAT6 cabling. CAT6A cables are fully backward compatible with previous categories, including CAT6 and 5e. Category 6A is also designed to support bundled cable installations up to 100 meters and PoE+ low-power implementations. The standard includes the performance parameter, Alien Crosstalk (ANEXT). Because of its higher performance transmission speeds and higher MHz rating, CAT6A cable needs to be tested for external noise outside the cable, which wasn’t a concern with previous cabling categories. CAT6A UTP also has a much larger diameter than previous cables.
Alien crosstalk (ANEXT) is a critical and unique measurement in 10-GbE systems. Crosstalk, measured in 10/100/1000BASE-T systems, is the mixing of signals between wire pairs within a cable. Alien Crosstalk, in 10-GbE systems, is the measurement of the unwanted signal coupling between wire pairs in different and adjacent cables or from one balanced twisted-pair component, channel, or permanent link to another.
The amount of ANEXT depends on a number of factors, including the type of cable, cable jacket, cable length, cable twist density, proximity of adjacent cables, and connectors, and EMI. Patch panels and connecting hardware are also affected by ANEXT.
With Alien Crosstalk, the affected cable is called the victim cable. The surrounding cables are the disturber cables.
There are a number of ways to mitigate the effects of ANEXT in CAT6A runs. According to the standards, ANEXT can be improved by laying CAT6A UTP cable loosely in pathways and raceways with space between the cables. This contrasts to the tightly bundled runs of CAT6/5e cable that we are used to. The tight bundles present a worst-case scenario of six cables around one, thus the center cable would be adversely affected by ANEXT. CAT6A UTP cable needs to be tested for ANEXT. This is a complex and time-consuming process in which all possible wire-pair combinations need to be tested for ANEXT and far-end ANEXT. It can take 50 minutes to test one link in a bundle of 24 CAT 6A UTP cables.
To virtually eliminate the problem of ANEXT, you can use CAT6A F/UTP cable. The F indicates an outer foil shield encasing four unshielded twisted pairs. This cable is also a good choice when security is an issue because it doesn’t emit signals. In addition, CAT6A F/UTP cable works well in noisy environments with a lot of EMI/RFI.
Installation of CAT6A F/UTP is simpler, too, because the cable features a smaller outside diameter than CAT6A UTP. Its construction makes it easier to pull and more resilient. The cable also has a smaller diameter so you can run more cables in a conduit or pathway, and have greater patch panel port density.
For more information, see the CAT6A F/UTP vs. UTP: What You Need to Know white paper in the Resources section at blackbox.com.
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Black Box Explains...10-GbE, CAT6A, and ANEXT.
The IEEE released the 802.3an 10GBASE-T standard in June 2006. This standard specifies 10-Gbps data transmission
over four-pair copper cabling. 10-Gigabit Ethernet (10-GbE) transmission includes up to 37 meters of CAT6 cable (with installation mitigation techniques), 100 meters of Augmented Category 6 (CAT6A) UTP or F/UTP cable or 100 meters
of S/FTP CAT7/Class F cable.
CAT6A is the ANSI/TIA 10-Gigabit Ethernet (10-GbE) over copper standard. Its requirements are covered in ANSI/TIA-568-C.2 (Balanced Twisted-Pair Communications Cabling and Components Standard) published in August 2009. It defines 10-Gigabit data transmission over a 4-connector twisted-pair CAT6A copper cable for a distance of 100 meters.
Category 6A cabling is designed to support next-generation applications, including the transfer of large amounts of data at high speeds, up to 10 Gbps. CAT6A extends electrical specifications to 500 MHz from 250 MHz for CAT6 cabling. CAT6A cables are fully backward compatible with previous categories, including CAT6 and 5e. Category 6A is also designed to support bundled cable installations up to 100 meters and PoE+ low-power implementations. The standard includes the performance parameter, Alien Crosstalk (ANEXT). Because of its higher performance transmission speeds and higher MHz rating, CAT6A cable needs to be tested for external noise outside the cable, which wasn’t a concern with previous cabling categories. CAT6A UTP also has a much larger diameter than previous cables.
Alien crosstalk (ANEXT) is a critical and unique measurement in 10-GbE systems. Crosstalk, measured in 10/100/1000BASE-T systems, is the mixing of signals between wire pairs within a cable. Alien Crosstalk, in 10-GbE systems, is the measurement of the unwanted signal coupling between wire pairs in different and adjacent cables or from one balanced twisted-pair component, channel, or permanent link to another.
The amount of ANEXT depends on a number of factors, including the type of cable, cable jacket, cable length, cable twist density, proximity of adjacent cables, and connectors, and EMI. Patch panels and connecting hardware are also affected by ANEXT.
With Alien Crosstalk, the affected cable is called the victim cable. The surrounding cables are the disturber cables.
There are a number of ways to mitigate the effects of ANEXT in CAT6A runs. According to the standards, ANEXT can be improved by laying CAT6A UTP cable loosely in pathways and raceways with space between the cables. This contrasts to the tightly bundled runs of CAT6/5e cable that we are used to. The tight bundles present a worst-case scenario of six cables around one, thus the center cable would be adversely affected by ANEXT. CAT6A UTP cable needs to be tested for ANEXT. This is a complex and time-consuming process in which all possible wire-pair combinations need to be tested for ANEXT and far-end ANEXT. It can take 50 minutes to test one link in a bundle of 24 CAT 6A UTP cables.
To virtually eliminate the problem of ANEXT, you can use CAT6A F/UTP cable. The F indicates an outer foil shield encasing four unshielded twisted pairs. This cable is also a good choice when security is an issue because it doesn’t emit signals. In addition, CAT6A F/UTP cable works well in noisy environments with a lot of EMI/RFI.
Installation of CAT6A F/UTP is simpler, too, because the cable features a smaller outside diameter than CAT6A UTP. Its construction makes it easier to pull and more resilient. The cable also has a smaller diameter so you can run more cables in a conduit or pathway, and have greater patch panel port density.
For more information, see the CAT6A F/UTP vs. UTP: What You Need to Know white paper in the Resources section at blackbox.com.
Black Box Explains...Token Ring Cabling
The original Token Ring specifications called for shielded twisted-pair (STP) cable using either a DB9 connector or a unique square connector called the IBM data connector. Later, Token Ring was... more/see it nowadapted to use conventional unshielded twisted-pair (UTP) cable with RJ-45 connectors. The most common kinds of Token Ring cabling in use to day are Type 1 and Type 6 STP as well as Type 3 UTP.
Type 1 shielded twisted-pair (STP) cable is the original wiring for Token Ring. In Type 1 cabling, each wire is constructed of 22 AWG solid copper. Type 1 cable is not as flexible as Type 6 cable and is generally used for long runs in areas where twists and turns are less likely, such as in walls or conduits.
Type 6 Token Ring cable is a lighter, more pliable version of Type 1 cable. It’s constructed of two stranded 26 AWG copper pairs that are surrounded by an overall braided shield. Type 6 cable is commonly used in offices and open areas, and its flexible construction enables it to negotiate multiple twists and turns.
Type 3 or UTP Token Ring cabling uses the same twisted-pair CAT3, CAT5, or CAT5e cabling with RJ-45 connectors as 10BASE-T Ethernet does. Attaching older Type 1 Token Ring to UTP Token Ring requires a balun or adapter. collapse
Black Box Explains...Token Ring Cabling
The original Token Ring specifications called for shielded twisted-pair (STP) cable using either a DB9 connector or a unique square connector called the IBM data connector. Later, Token Ring was adapted to use conventional unshielded twisted-pair (UTP) cable with RJ-45 connectors. The most common kinds of Token Ring cabling in use to day are Type 1 and Type 6 STP as well as Type 3 UTP.
Type 1 shielded twisted-pair (STP) cable is the original wiring for Token Ring. In Type 1 cabling, each wire is constructed of 22 AWG solid copper. Type 1 cable is not as flexible as Type 6 cable and is generally used for long runs in areas where twists and turns are less likely, such as in walls or conduits.
Type 6 Token Ring cable is a lighter, more pliable version of Type 1 cable. It’s constructed of two stranded 26 AWG copper pairs that are surrounded by an overall braided shield. Type 6 cable is commonly used in offices and open areas, and its flexible construction enables it to negotiate multiple twists and turns.
Type 3 or UTP Token Ring cabling uses the same twisted-pair CAT3, CAT5, or CAT5e cabling with RJ-45 connectors as 10BASE-T Ethernet does. Attaching older Type 1 Token Ring to UTP Token Ring requires a balun or adapter.
Black Box Explains…VoIP
Voice over Internet Protocol (VoIP) is a recently developed, cost-saving alternative to traditional telephone service that enables voice data to be transported over IP networks, like the Internet, instead of... more/see it nowthe public switched telephone network (PSTN) or a cellular network.
VoIP, which operates strictly over IP networks, can connect to other VoIP nodes or traditional phone lines. The IP network used may be the Internet or a private network.
In either instance, the actual data-transport portion of this network can still be made up of the full gamut of network services: high-speed leased lines, Frame Relay, ATM, DSL, copper, fiber, wireless, satellite, and microwave signals. VoIP simply digitizes voice data and adds it to other information traveling along the same network.
With this flexible technology, a phone call can be placed between two PCs, between a PC and a standard telephone, between a PC and an IP phone, between
an IP phone and a standard telephone, or between two IP phones. It will take a long time for the PSTN to support this technology seamlessly, but this seems to be the direction in which phone systems are headed.
Benefits of VoIP
Because VoIP is inexpensive, has a worldwide reach, and operates on a few simple principles, it’s exploded in popularity in recent years—especially among both small and large businesses that incur significant long-distance telephone expenses.
Savings
Without question, the primary benefit
of a VoIP system is decreasing or eliminating long-distance telephone charges. Organizations with a high volume of long-distance voice traffic stand to save quite a lot of money by implementing a VoIP system. However, this factor alone may not warrant a full commitment to VoIP for some companies.
Setup fees for VoIP are usually quite low so your organization can generally start saving money after only a month or two of service. And with the wide variety of VoIP products and services on the market, it’s easier than ever to set up a VoIP phone system over your network.
Convenience
VoIP can be set up in a way that enables you to use phone numbers in exactly the same way as you did before VoIP. Most of the services you get with traditional phone service—Voice Mail, Call Waiting, and Call Routing, for instance—are also available with VoIP.
VoIP doesn’t interfere with other network services either, so you can surf the Web while making a VoIP call.
Portability
VoIP doesn’t tie you to one phone or to a single location. Anywhere you find high-speed reliable Internet access, you can use VoIP. Your phone number stays the same wherever you are—office, home, hotel,
or even traveling overseas.
Standards
Although the ITU standards for VoIP have evolved significantly in the last few years, VoIP is still suffering from a lack of generally accepted interoperability standards.
H.323, a standard for real-time audio, video, and data communications across IP-based networks (including the Internet), is almost universally accepted as the primary standard for VoIP call setup and signaling. It’s actually a collection of standards that works together for sending multimedia and data over networks that don’t provide guaranteed Quality of Service (QoS).
The H.323 standard includes:
- Real-Time Transport Protocol (RTP) specifies end-to-end network transport functions for applications transmitting real-time data such as video. RTP provides services like payload type identification, sequence numbering, time stamping, and delivery monitoring to real-time applications. Plus, it works with RTCP.
- Real-time Transport Control Protocol (RTCP) works with RTP to provide a feedback mechanism, providing QoS status and control information to the streaming server.
- Registration, Admission, Status (RAS) is a gateway protocol that manages functions such as signaling, registration, admissions, bandwidth changes, status, and disengage procedures.
- Q.931 manages call setup and termination.
- H.245 negotiates channel usage and capabilities.
- H.235 provides security and authentication.
As VoIP product manufacturers began conducting interoperability tests for more complex operations, they recognized that they needed a simpler and more adaptable standard for call handling and signaling protocol.
To this end, the IETF developed the Session Initiation Protocol (SIP). SIP is built with less computer code than H.323 is, so it’s less cumbersome. Because SIP is similar in nature to HTML—it uses ASCII text for configuration—users can adapt it more easily for specific VoIP systems. In contrast, modifying H.323 for VoIP applications requires a knowledgeable computer programmer.
Both H.323 and SIP are considered “thick clients,” where intelligence is maintained in the end devices such as IP telephones. In this respect, H.323 has a head start, although most VoIP systems today support both H.323 and SIP.
Providers
Despite the fact that VoIP standards are still developing, providers are already flooding the market with products and services while forming partnerships and matching expertise to strengthen their position in this new market. The biggest of these players and alliances—the ones who have the size and experience to grasp technical issues and quickly build infrastructures over which to offer VoIP services—are able to keep up with (and often influence) the continual changes in this market and keep rolling out new services.
Components
A VoIP system depends on devices that connect your traditional phone or phone system to an IP network. Components that you’ll see in a VoIP system include:
- End-user devices
- Gateways or gatekeepers
- IPBXs
- IP Networks
End-user devices are usually VoIP telephones or PCs running VoIP software. End-user devices have their own IP address and make a direct connection to the IP network.
A gateway is a device that converts circuit-switched analog voice calls from a traditional PBX into VoIP packets and transmits them over an IP network either
to another gateway or directly to an end-user device.
A gateway can have additional features such as voice compression, echo cancellation, and packet prioritization.
Because VoIP-enabled end-user devices can communicate directly with each other over an IP network, a gateway is not a required component of a VoIP system as long as the VoIP devices are connected directly to the IP network.
An IPBX is a PBX with a built-in gateway. IPBX systems are equipped for hundreds of telephone ports, with WAN support for trunk connections to the PSTN, and with high-speed IP WAN links. In addition to VoIP features, these systems usually include other features typical of traditional PBX systems such as music on hold, auto-attendant, and call management. Often, they include Ethernet ports to support VoIP telephones.
VoIP can be set up with or without a connection to standard PSTN phone service. You can, of course, place calls over the Internet directly from your PC or IP phone to another VoIP-enabled device. But what makes VoIP so versatile is that, through the use of a gateway service, it can also be used to call the numbers of phones connected to standard land-line or cellular phone services. They can also receive calls from standard telephones.
Not all fun and free calls
There are still things to consider when you’re deciding whether or not to invest in VoIP.
Regulation vagaries
Much of the government regulation of VoIP is still being worked out. The U.S. government hasn’t decided whether VoIP is going to be regulated as phone service
or whether to tax it. VoIP isn’t available worldwide because some governments fear the loss of tax revenue or control.
Compatibility
Although older VoIP equipment may still have some compatibility issues, current VoIP products from different vendors generally work together.
Cost
For all the popular talk about VoIP being free, it isn’t truly free. Any VoIP system has costs associated with its implementation—equipment, high-speed Internet access, and gateway service. So, although it’s inexpensive, it’s a long way from being free. For organizations with a high volume of long-distance calls, especially to international locations, VoIP almost always pays for itself quickly. However, private users or organizations with a low volume of long-distance calls primarily within the U.S., may find that a standard service is actually more economical in the short- to mid-term.
QoS
VoIP depends on having a fast, reliable network to operate. A fast network connection with guaranteed bandwidth is not a problem in a corporate intranet where you have complete control over
the network. However, if you’re using the Internet for VoIP, you’re using a public network that may be subject to slowdowns that cause drop-outs and distortion. You may find that your high-speed Internet connection is faster than the actual Internet and that the quality of your connection is generally unacceptable or is unacceptable at times when Internet usage is high.
There are four common network issues that can cause problems with a VoIP system:
- Latency is a delay in data transmission. With VoIP, this usually results in people speaking over one another because neither can tell when the other is finished talking.
- Loss. Losing a small percentage of voice transmission doesn’t affect VoIP, but too much (more than 1%) compromises the quality of the call.
- Jitter—is common to congested networks with bursty traffic. Jitter can be managed to some degree with software buffers.
- Sequence errors—or changes in the order of packets when they’re recompiled at the receiving station, degrades sound quality.
Emergency services
If you subscribe to a VoIP gateway service that enables you to use your VoIP phone like a regular phone, be aware that you may not be able to call 911 for emergencies. If 911 service is important to you because you don’t have an alternative way to call 911, shop for a VoIP provider who does provide this service.
Consider, too, that VoIP needs both working Internet access and power to work. If you lose your Internet service, your phone goes, too. And, unlike regular phone service that can keep basic telephones working when the power goes out, VoIP needs power—if you lose power, you lose your phone.
Moving forward
Before VoIP technology becomes truly universal, the current worldwide PSTN will have to migrate to a packet-based IP equivalent. Industry inertia alone dictates this will not occur instantly. The current worldwide PSTN system has grown to what it is over a period of 125 years. Given the sheer complexity of the existing PSTN, the migration to an IP packet network will probably occur during several decades.
As migration from the PSTN to IP-based networks proceeds, businesses and home users will gradually discover reasons of their own to implement VoIP. It won’t happen right away, but we predict that VoIP will become a big part of telecommunications
in the not-so-distant future.
Although it’s not quite as convenient as conventional phone service, VoIP can offer serious savings—particularly if you now regularly pay for multiple overseas phone calls. Keep in mind though, VoIP isn’t a one-size-fits-all solution. But with a little planning, VoIP could spell savings for you! collapse
Black Box Explains…VoIP
Voice over Internet Protocol (VoIP) is a recently developed, cost-saving alternative to traditional telephone service that enables voice data to be transported over IP networks, like the Internet, instead of the public switched telephone network (PSTN) or a cellular network.
VoIP, which operates strictly over IP networks, can connect to other VoIP nodes or traditional phone lines. The IP network used may be the Internet or a private network.
In either instance, the actual data-transport portion of this network can still be made up of the full gamut of network services: high-speed leased lines, Frame Relay, ATM, DSL, copper, fiber, wireless, satellite, and microwave signals. VoIP simply digitizes voice data and adds it to other information traveling along the same network.
With this flexible technology, a phone call can be placed between two PCs, between a PC and a standard telephone, between a PC and an IP phone, between
an IP phone and a standard telephone, or between two IP phones. It will take a long time for the PSTN to support this technology seamlessly, but this seems to be the direction in which phone systems are headed.
Benefits of VoIP
Because VoIP is inexpensive, has a worldwide reach, and operates on a few simple principles, it’s exploded in popularity in recent years—especially among both small and large businesses that incur significant long-distance telephone expenses.
Savings
Without question, the primary benefit
of a VoIP system is decreasing or eliminating long-distance telephone charges. Organizations with a high volume of long-distance voice traffic stand to save quite a lot of money by implementing a VoIP system. However, this factor alone may not warrant a full commitment to VoIP for some companies.
Setup fees for VoIP are usually quite low so your organization can generally start saving money after only a month or two of service. And with the wide variety of VoIP products and services on the market, it’s easier than ever to set up a VoIP phone system over your network.
Convenience
VoIP can be set up in a way that enables you to use phone numbers in exactly the same way as you did before VoIP. Most of the services you get with traditional phone service—Voice Mail, Call Waiting, and Call Routing, for instance—are also available with VoIP.
VoIP doesn’t interfere with other network services either, so you can surf the Web while making a VoIP call.
Portability
VoIP doesn’t tie you to one phone or to a single location. Anywhere you find high-speed reliable Internet access, you can use VoIP. Your phone number stays the same wherever you are—office, home, hotel,
or even traveling overseas.
Standards
Although the ITU standards for VoIP have evolved significantly in the last few years, VoIP is still suffering from a lack of generally accepted interoperability standards.
H.323, a standard for real-time audio, video, and data communications across IP-based networks (including the Internet), is almost universally accepted as the primary standard for VoIP call setup and signaling. It’s actually a collection of standards that works together for sending multimedia and data over networks that don’t provide guaranteed Quality of Service (QoS).
The H.323 standard includes:
- Real-Time Transport Protocol (RTP) specifies end-to-end network transport functions for applications transmitting real-time data such as video. RTP provides services like payload type identification, sequence numbering, time stamping, and delivery monitoring to real-time applications. Plus, it works with RTCP.
- Real-time Transport Control Protocol (RTCP) works with RTP to provide a feedback mechanism, providing QoS status and control information to the streaming server.
- Registration, Admission, Status (RAS) is a gateway protocol that manages functions such as signaling, registration, admissions, bandwidth changes, status, and disengage procedures.
- Q.931 manages call setup and termination.
- H.245 negotiates channel usage and capabilities.
- H.235 provides security and authentication.
As VoIP product manufacturers began conducting interoperability tests for more complex operations, they recognized that they needed a simpler and more adaptable standard for call handling and signaling protocol.
To this end, the IETF developed the Session Initiation Protocol (SIP). SIP is built with less computer code than H.323 is, so it’s less cumbersome. Because SIP is similar in nature to HTML—it uses ASCII text for configuration—users can adapt it more easily for specific VoIP systems. In contrast, modifying H.323 for VoIP applications requires a knowledgeable computer programmer.
Both H.323 and SIP are considered “thick clients,” where intelligence is maintained in the end devices such as IP telephones. In this respect, H.323 has a head start, although most VoIP systems today support both H.323 and SIP.
Providers
Despite the fact that VoIP standards are still developing, providers are already flooding the market with products and services while forming partnerships and matching expertise to strengthen their position in this new market. The biggest of these players and alliances—the ones who have the size and experience to grasp technical issues and quickly build infrastructures over which to offer VoIP services—are able to keep up with (and often influence) the continual changes in this market and keep rolling out new services.
Components
A VoIP system depends on devices that connect your traditional phone or phone system to an IP network. Components that you’ll see in a VoIP system include:
- End-user devices
- Gateways or gatekeepers
- IPBXs
- IP Networks
End-user devices are usually VoIP telephones or PCs running VoIP software. End-user devices have their own IP address and make a direct connection to the IP network.
A gateway is a device that converts circuit-switched analog voice calls from a traditional PBX into VoIP packets and transmits them over an IP network either
to another gateway or directly to an end-user device.
A gateway can have additional features such as voice compression, echo cancellation, and packet prioritization.
Because VoIP-enabled end-user devices can communicate directly with each other over an IP network, a gateway is not a required component of a VoIP system as long as the VoIP devices are connected directly to the IP network.
An IPBX is a PBX with a built-in gateway. IPBX systems are equipped for hundreds of telephone ports, with WAN support for trunk connections to the PSTN, and with high-speed IP WAN links. In addition to VoIP features, these systems usually include other features typical of traditional PBX systems such as music on hold, auto-attendant, and call management. Often, they include Ethernet ports to support VoIP telephones.
VoIP can be set up with or without a connection to standard PSTN phone service. You can, of course, place calls over the Internet directly from your PC or IP phone to another VoIP-enabled device. But what makes VoIP so versatile is that, through the use of a gateway service, it can also be used to call the numbers of phones connected to standard land-line or cellular phone services. They can also receive calls from standard telephones.
Not all fun and free calls
There are still things to consider when you’re deciding whether or not to invest in VoIP.
Regulation vagaries
Much of the government regulation of VoIP is still being worked out. The U.S. government hasn’t decided whether VoIP is going to be regulated as phone service
or whether to tax it. VoIP isn’t available worldwide because some governments fear the loss of tax revenue or control.
Compatibility
Although older VoIP equipment may still have some compatibility issues, current VoIP products from different vendors generally work together.
Cost
For all the popular talk about VoIP being free, it isn’t truly free. Any VoIP system has costs associated with its implementation—equipment, high-speed Internet access, and gateway service. So, although it’s inexpensive, it’s a long way from being free. For organizations with a high volume of long-distance calls, especially to international locations, VoIP almost always pays for itself quickly. However, private users or organizations with a low volume of long-distance calls primarily within the U.S., may find that a standard service is actually more economical in the short- to mid-term.
QoS
VoIP depends on having a fast, reliable network to operate. A fast network connection with guaranteed bandwidth is not a problem in a corporate intranet where you have complete control over
the network. However, if you’re using the Internet for VoIP, you’re using a public network that may be subject to slowdowns that cause drop-outs and distortion. You may find that your high-speed Internet connection is faster than the actual Internet and that the quality of your connection is generally unacceptable or is unacceptable at times when Internet usage is high.
There are four common network issues that can cause problems with a VoIP system:
- Latency is a delay in data transmission. With VoIP, this usually results in people speaking over one another because neither can tell when the other is finished talking.
- Loss. Losing a small percentage of voice transmission doesn’t affect VoIP, but too much (more than 1%) compromises the quality of the call.
- Jitter—is common to congested networks with bursty traffic. Jitter can be managed to some degree with software buffers.
- Sequence errors—or changes in the order of packets when they’re recompiled at the receiving station, degrades sound quality.
Emergency services
If you subscribe to a VoIP gateway service that enables you to use your VoIP phone like a regular phone, be aware that you may not be able to call 911 for emergencies. If 911 service is important to you because you don’t have an alternative way to call 911, shop for a VoIP provider who does provide this service.
Consider, too, that VoIP needs both working Internet access and power to work. If you lose your Internet service, your phone goes, too. And, unlike regular phone service that can keep basic telephones working when the power goes out, VoIP needs power—if you lose power, you lose your phone.
Moving forward
Before VoIP technology becomes truly universal, the current worldwide PSTN will have to migrate to a packet-based IP equivalent. Industry inertia alone dictates this will not occur instantly. The current worldwide PSTN system has grown to what it is over a period of 125 years. Given the sheer complexity of the existing PSTN, the migration to an IP packet network will probably occur during several decades.
As migration from the PSTN to IP-based networks proceeds, businesses and home users will gradually discover reasons of their own to implement VoIP. It won’t happen right away, but we predict that VoIP will become a big part of telecommunications
in the not-so-distant future.
Although it’s not quite as convenient as conventional phone service, VoIP can offer serious savings—particularly if you now regularly pay for multiple overseas phone calls. Keep in mind though, VoIP isn’t a one-size-fits-all solution. But with a little planning, VoIP could spell savings for you!
Black Box Explains...Fiber optic attenuators.
Attenuators are used with single-mode fiber optic devices and cable to filter the strength of the fiber optic signal. Depending on the type of attenuator attached to the devices at... more/see it noweach end of the fiber optic cable, you can diminish the strength of the light signal a variable amount, measured in decibels (dB).
Why would you want to filter the strength of the fiber optic signal? Single-mode fiber is designed to carry a fiber optic signal long distances—as much as 70 kilometers (or 43.4 miles). Fiber devices send this signal with great force to ensure that the signal, and your data, arrive at the other end intact.
But when two fiber devices connected with single-mode fiber cable are close to each other, the signal may be too strong. As a result, the light signal reflects back down the fiber cable. Data can be corrupted and transmissions can be faulty. A signal that is too strong can even damage the attached equipment.
Because its probably not feasible to move your fiber equipment farther apart, the easiest solution is to attach an attenuator to each fiber device. Just as sunglasses filter the strength of sunlight, attenuators filter the strength of the light signal transmitted along single-mode fiber cable. Within the attenuator, theres doping that reduces the strength of the signal passing through the fiber connection and minute air gaps where the two fibers meet. Fiber grooves may also be intentionally misaligned by several microns—but only enough to slow the fiber optic signal to an acceptable rate as it travels down the cable.
Before selecting an attenuator, you need to check the type of adapter on your fiber devices. Attenuators typically fit into any patch panel equipped with FC, SC, or LC adapters that contain either PC or APC contacts. In addition to the type of adapter, you also need to determine the necessary attenuation value, such as 5 or 10 dB. This value varies, depending on the strength of fiber optic signal desired. collapse
Black Box Explains...Fiber optic attenuators.
Attenuators are used with single-mode fiber optic devices and cable to filter the strength of the fiber optic signal. Depending on the type of attenuator attached to the devices at each end of the fiber optic cable, you can diminish the strength of the light signal a variable amount, measured in decibels (dB).
Why would you want to filter the strength of the fiber optic signal? Single-mode fiber is designed to carry a fiber optic signal long distances—as much as 70 kilometers (or 43.4 miles). Fiber devices send this signal with great force to ensure that the signal, and your data, arrive at the other end intact.
But when two fiber devices connected with single-mode fiber cable are close to each other, the signal may be too strong. As a result, the light signal reflects back down the fiber cable. Data can be corrupted and transmissions can be faulty. A signal that is too strong can even damage the attached equipment.
Because its probably not feasible to move your fiber equipment farther apart, the easiest solution is to attach an attenuator to each fiber device. Just as sunglasses filter the strength of sunlight, attenuators filter the strength of the light signal transmitted along single-mode fiber cable. Within the attenuator, theres doping that reduces the strength of the signal passing through the fiber connection and minute air gaps where the two fibers meet. Fiber grooves may also be intentionally misaligned by several microns—but only enough to slow the fiber optic signal to an acceptable rate as it travels down the cable.
Before selecting an attenuator, you need to check the type of adapter on your fiber devices. Attenuators typically fit into any patch panel equipped with FC, SC, or LC adapters that contain either PC or APC contacts. In addition to the type of adapter, you also need to determine the necessary attenuation value, such as 5 or 10 dB. This value varies, depending on the strength of fiber optic signal desired.
Black Box Explains...The 13W3 connector.
The 13W3 connector, also called a 13C3 or DB13W3 connector, is an unusual connector that combines a 10-pin D-shell with three analog video conductors. It supports very-high-resolution analog video signals... more/see it nowand has been used by Sun Microsystems®, SGI, NeXt, Intergraph, and other manufacturers. Although 13W3 connectors from different manufacturers look the same, they may be pinned differently.
Pinning for a standard Sun® 13W3 connector:
A1: Red
A2: Green/Gray
A3: Blue
1: Ground*
2: Vertical Sync*
3: Sense 2
4: Sense Ground
5: Composite Sync
6: Horizontal Sync*
7: Ground*
8: Sense 1
9: Sense 0
10: Composite Ground
* Considered obsolete; may not be connected. collapse
Black Box Explains...The 13W3 connector.
The 13W3 connector, also called a 13C3 or DB13W3 connector, is an unusual connector that combines a 10-pin D-shell with three analog video conductors. It supports very-high-resolution analog video signals and has been used by Sun Microsystems®, SGI, NeXt, Intergraph, and other manufacturers. Although 13W3 connectors from different manufacturers look the same, they may be pinned differently.
Pinning for a standard Sun® 13W3 connector:
A1: Red
A2: Green/Gray
A3: Blue
1: Ground*
2: Vertical Sync*
3: Sense 2
4: Sense Ground
5: Composite Sync
6: Horizontal Sync*
7: Ground*
8: Sense 1
9: Sense 0
10: Composite Ground
* Considered obsolete; may not be connected.
Black Box Explains... Speaker wire gauge.
Wire gauge (often shown as AWG, for American Wire Gauge) is a measure of the thickness of the wire. The more a wire is drawn or sized, the smaller its... more/see it nowdiameter will be. The lower the wire gauge, the thicker the wire.
For example, a 24 AWG wire is thinner than a 14 AWG wire. A lower AWG means longer transmission distance and better integrity. As a rule of thumb, power loss decreases as the wire size increases.
When it comes to choosing speaker cable, consider a few factors: distance, the type of system and amplifier you have, the frequencies of the signals being handled, and any specifications that the speaker manufacturer recommends.
For most home applications where you simply need to run cable from your stereo to speakers in the same room—or even behind the walls to other rooms—16 AWG cable is usually fine.
If youre considering runs of more than 40 feet (12.1 m), consider using 14 AWG or even 12 AWG cable. They both offer better transmission and less resistance over longer distances. You should probably choose 12 AWG cable for high-end audio systems with higher power output or for low-frequency subwoofers. As a rule of thumb, power loss decreases as the wire size increases.
To terminate your cable, choose gold connectors. Because gold resists oxidation over time, gold connectors wear better and offer better peformance than other connectors do. collapse
Black Box Explains... Speaker wire gauge.
Wire gauge (often shown as AWG, for American Wire Gauge) is a measure of the thickness of the wire. The more a wire is drawn or sized, the smaller its diameter will be. The lower the wire gauge, the thicker the wire.
For example, a 24 AWG wire is thinner than a 14 AWG wire. A lower AWG means longer transmission distance and better integrity. As a rule of thumb, power loss decreases as the wire size increases.
When it comes to choosing speaker cable, consider a few factors: distance, the type of system and amplifier you have, the frequencies of the signals being handled, and any specifications that the speaker manufacturer recommends.
For most home applications where you simply need to run cable from your stereo to speakers in the same room—or even behind the walls to other rooms—16 AWG cable is usually fine.
If youre considering runs of more than 40 feet (12.1 m), consider using 14 AWG or even 12 AWG cable. They both offer better transmission and less resistance over longer distances. You should probably choose 12 AWG cable for high-end audio systems with higher power output or for low-frequency subwoofers. As a rule of thumb, power loss decreases as the wire size increases.
To terminate your cable, choose gold connectors. Because gold resists oxidation over time, gold connectors wear better and offer better peformance than other connectors do.
Black Box Explains...Beyond T1—other standards for high-speed circuits.
While there are many applications for basic T1 rate service (1.536 Mbps), some applications require much more bandwidth. One of the most attractive features of T1 is the number of... more/see it nowoptions available to accommodate these kinds of demands. The important thing to remember is that all of these higher-speed services operate with the same consistent framing formats as the standard T1 service.
T1 is a high-speed service with a clock speed of 1.544 Mbps. It’s made up of 24 64-kbps DS0 (Digital-Signal [zero]) subchannels that together can support throughput rates of up to 1.536 Mbps. But there are higher levels of T1 service that are also available. For instance, T1C service doubles the T1 rate. It supports 3.152 Mbps with a total of 48 DS0s for top-speed applications. In a T1C environment, two T1 lines are combined into one using a special T1 mux.
The next-highest level of service is called T2. It offers 6.312 Mbps over 96 DS0s by multi-plexing 4 T1 lines into a single high-speed line.
The next two levels of service are exponentially larger than T2. A high-speed T3 trunk line is 28 times larger than a standard T1 line. T3 brings 44.736 Mbps to a customer site via 672 DS0s. This tremendous capacity is made possible by multiplexing 28 T1 lines or combina?tions of T2 and T1 lines.
Finally, T4 service offers a bandwidth potential of 274.176 Mbps, made up of 4032 64-kbps DS0 subchannels. At 168 times the size of a standard 1.544-Mbps line, T4 service dwarfs T1. The physical connections require multiplexing 6 T3 lines or 168 T1 lines into a single high-speed trunk.
With so many incredibly high-speed T-level service options available, system administrators have great flexibility to configure their operations for maximum efficiency and economy.
It’s important to remember that these various levels of T1 services can be implemented simultaneously within a particularly large enterprise to support complex network configurations.
Of course, this kind of application has the potential to become somewhat overwhelming from a management standpoint. However, as long as you keep track of the individual DS0s, you should always be able to accurately gauge how much available bandwidth you have at your disposal. collapse
Black Box Explains...Beyond T1—other standards for high-speed circuits.
While there are many applications for basic T1 rate service (1.536 Mbps), some applications require much more bandwidth. One of the most attractive features of T1 is the number of options available to accommodate these kinds of demands. The important thing to remember is that all of these higher-speed services operate with the same consistent framing formats as the standard T1 service.
T1 is a high-speed service with a clock speed of 1.544 Mbps. It’s made up of 24 64-kbps DS0 (Digital-Signal [zero]) subchannels that together can support throughput rates of up to 1.536 Mbps. But there are higher levels of T1 service that are also available. For instance, T1C service doubles the T1 rate. It supports 3.152 Mbps with a total of 48 DS0s for top-speed applications. In a T1C environment, two T1 lines are combined into one using a special T1 mux.
The next-highest level of service is called T2. It offers 6.312 Mbps over 96 DS0s by multi-plexing 4 T1 lines into a single high-speed line.
The next two levels of service are exponentially larger than T2. A high-speed T3 trunk line is 28 times larger than a standard T1 line. T3 brings 44.736 Mbps to a customer site via 672 DS0s. This tremendous capacity is made possible by multiplexing 28 T1 lines or combina?tions of T2 and T1 lines.
Finally, T4 service offers a bandwidth potential of 274.176 Mbps, made up of 4032 64-kbps DS0 subchannels. At 168 times the size of a standard 1.544-Mbps line, T4 service dwarfs T1. The physical connections require multiplexing 6 T3 lines or 168 T1 lines into a single high-speed trunk.
With so many incredibly high-speed T-level service options available, system administrators have great flexibility to configure their operations for maximum efficiency and economy.
It’s important to remember that these various levels of T1 services can be implemented simultaneously within a particularly large enterprise to support complex network configurations.
Of course, this kind of application has the potential to become somewhat overwhelming from a management standpoint. However, as long as you keep track of the individual DS0s, you should always be able to accurately gauge how much available bandwidth you have at your disposal.
Black Box Explains...The fully accessorized rack.
After you choose your rack, consider how youll set it up and what accessories you might need.
Your rack may need to be secured. A typical rack has about a... more/see it now15"-deep base, providing some stability, but not enough to prevent the rack from tipping if heavy objects are mounted on it. To solve this problem, most rack bases can be bolted to the floor.
You also need to decide how to accommodate standalone equipment, which is not actually rackmounted or bolted to the rack. You can place small devices on a cantilevered shelf such as the RM001, however, you should place heavier items such as monitors on a center-weight shelf such as the RM377.
Small extras, such as Patch Panel Hinge Kits, can make your job easier. These hinges enable you to access the back of a patch panel simply by swinging it out from the rack. Theyre particularly useful for racks in hard-to-reach areas.
If you need to mount both 19" and 23" equipment in the same rack, use a 23" rack with 23"-to-19" Rackmount Adapters to fit the 19" devices.
For a neater appearance, you can cover unused spaces in a rack with Filler Panels.
Cable management is also an important consideration. Our Horizontal and Vertical Cable Managers help you to route cables along the sides of racks, between racks, and to the rackmounted equipment. collapse
Black Box Explains...The fully accessorized rack.
After you choose your rack, consider how youll set it up and what accessories you might need.
Your rack may need to be secured. A typical rack has about a 15"-deep base, providing some stability, but not enough to prevent the rack from tipping if heavy objects are mounted on it. To solve this problem, most rack bases can be bolted to the floor.
You also need to decide how to accommodate standalone equipment, which is not actually rackmounted or bolted to the rack. You can place small devices on a cantilevered shelf such as the RM001, however, you should place heavier items such as monitors on a center-weight shelf such as the RM377.
Small extras, such as Patch Panel Hinge Kits, can make your job easier. These hinges enable you to access the back of a patch panel simply by swinging it out from the rack. Theyre particularly useful for racks in hard-to-reach areas.
If you need to mount both 19" and 23" equipment in the same rack, use a 23" rack with 23"-to-19" Rackmount Adapters to fit the 19" devices.
For a neater appearance, you can cover unused spaces in a rack with Filler Panels.
Cable management is also an important consideration. Our Horizontal and Vertical Cable Managers help you to route cables along the sides of racks, between racks, and to the rackmounted equipment.
Black Box Explains... Digital Visual Interface (DVI).
The Digital Visual Interface (DVI) video standard is based on transition-minimized differential signaling (TMDS). In a typical single-line digital signal, voltage is raised to a high level and decreased to... more/see it nowa low level to create transitions that convey data. To minimize the number of tran-sitions needed to transfer data, TMDS
uses a pair of signal wires. When one wire goes to a high-voltage state, the other goes to a low-voltage state. This balance increases the data-transfer rate and improves accuracy.
Although there are four types of DVI connectors, only DVI-D and DVI-I are commonly used for monitors. DVI-D is a digital-only connector. DVI-I supports both digital and analog RGB connections. collapse
Black Box Explains... Digital Visual Interface (DVI).
The Digital Visual Interface (DVI) video standard is based on transition-minimized differential signaling (TMDS). In a typical single-line digital signal, voltage is raised to a high level and decreased to a low level to create transitions that convey data. To minimize the number of tran-sitions needed to transfer data, TMDS
uses a pair of signal wires. When one wire goes to a high-voltage state, the other goes to a low-voltage state. This balance increases the data-transfer rate and improves accuracy.
Although there are four types of DVI connectors, only DVI-D and DVI-I are commonly used for monitors. DVI-D is a digital-only connector. DVI-I supports both digital and analog RGB connections.
Black Box Explains...Microphone positioning.
Proper microphone positioning is especially important to take advantage of noise canceling microphones, which reject background noise.
For optimum performance, position the microphone one finger width away from your lower lip.
Black Box Explains...Microphone positioning.
Proper microphone positioning is especially important to take advantage of noise canceling microphones, which reject background noise.
For optimum performance, position the microphone one finger width away from your lower lip.