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Test Code : 300-075
Test Name : Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2)
Vendor Name : Cisco
: 649 Real Questions

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Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) book

ebook Extract: implementing Cisco Unified Communications | 300-075 Real Questions and VCE Practice Test

Chapter 1, choosing issues in a Multisite Deployment

Deploying Cisco Unified Communications manager between multiple sites requires a suitable dial plan, acceptable bandwidth, a top quality of service (QoS) implementation and a design that may live on IP WAN failures.

This chapter will clarify the considerations that may arise in a multisite Cisco UC supervisor deployment, and advocate selected options.

identifying considerations in a Multisite DeploymentTable of contents:1. Multisite deployment problem overview2. first-class challenges3. Bandwidth challenges4. Availability challenges5. Dial plan challenges6. Overlapping and nonconsecutive numbers7. mounted versus variable-size numbering plans8. Variable-length numbering, E.164 addressing and DID9. Optimized name routing and PSTN backup10. NAT and safety considerations

Multisite deployment problem overview

previously known as Cisco call manager (CCM), Cisco Unified Communications supervisor (CUCM) multi-site deployment challenges can encompass the following:

  • high-quality considerations: precise-time communications of voice and video need to be prioritized over a packet-switching network. All site visitors is treated equally by default in routers and switches. Voice and video are delay-sensitive packets that need to be given priority to prevent extend and jitter (variable delay), which might result in lowered voice first-class.
  • Bandwidth considerations: Cisco Unified Communications (Cisco UC) can include voice and video streams, signaling site visitors, administration traffic, and utility traffic corresponding to rich media conferencing. The extra bandwidth it's required when deploying a Cisco Unified Communications solution needs to be calculated and provisioned for to make certain that information purposes and Cisco Unified Communications applications don't overload the available bandwidth. Bandwidth reservations will also be made to functions through QoS deployment.
  • Availability issues: When deploying Cisco Unified Communications manager (CUCM) with centralized name processing, IP telephones register with CUCM over the IP LAN and doubtlessly over the WAN. If gateways in far flung sites are the use of Media Gateway manage Protocol (MGCP) as a signaling protocol, they additionally depend upon the availability of CUCM appearing as an MGCP call agent. it is critical to put in force fallback options for IP phones and gateways in eventualities during which the connection to the CUCM servers is damaged as a result of IP WAN failure. Fallback solutions additionally follow to H.323 gateways but are already created with H.323 dial peers in a proper H.323 gateway configuration.
  • Dial plan considerations: directory numbers (DN) can overlap throughout numerous sites. Overlapping dial plans and nonconsecutive numbers may also be solved via designing a strong multisite dial plan. prevent overlapping numbers across websites each time viable for a simpler design.
  • NAT and security concerns: the use of inner most IP addresses inside an business IP community is terribly typical. information superhighway Telephony carrier providers (ITSP) require entertaining public IP addresses to route IP mobile calls. The private IP addresses in the business have to be translated into public IP addresses. Public IP addresses make the IP phones seen from the internet and therefore area to assaults.
  • high-quality challenges

    IP networks were no longer at the beginning designed to carry precise-time site visitors; as a substitute, they had been designed for resiliency and fault tolerance. every packet is processed separately in an IP network, every now and then causing distinctive packets in a communications circulation to take diverse paths to the vacation spot. The different paths in the community may also have a distinct amount of packet loss, prolong, and prolong version (jitter) on account of bandwidth, distance, and congestion modifications. The destination ought to be able to get hold of packets out of order and resequence these packets. This challenge is solved by means of precise-Time Transport Protocol (RTP) sequence numbers and site visitors resequencing. When feasible, it's most fulfilling to not depend fully on these RTP mechanisms. correct network design, the usage of Cisco router Cisco specific Forwarding (CEF) change cache technology, performs per-vacation spot load sharing by default. Per-vacation spot load sharing isn't a perfect load-balancing paradigm, however ensures that each and every IP movement (voice name) takes the equal path.

    Bandwidth is shared by using dissimilar clients and functions, whereas the amount of bandwidth required for someone IP flow varies vastly throughout short lapses of time. Most records purposes are very bursty, whereas Cisco precise-time audio communications with RTP use the equal continual-bandwidth movement. The bandwidth available for any software, together with CUCM and voice-bearer traffic, is unpredictable. all the way through height intervals, packets should be buffered in queues ready to be processed as a result of network congestion. Queuing is a term that any individual who has ever experienced air flight is accepted with. should you arrive on the airport, you need to get in a line (queue), since the number of ticket agents (bandwidth) accessible to investigate you in is less than the circulation of site visitors arriving at the ticket counters (incoming IP site visitors). If congestion occurs for too long, the queue (packet buffers) gets stuffed up, and passengers are irritated (packets are dropped). larger queuing delays and packet drops are more possible on tremendously loaded, sluggish-velocity links such as WAN links used between sites in a multisite environment. fine challenges are common on these forms of hyperlinks, and you should handle them by enforcing QoS. devoid of the use of QoS, voice packets experience lengthen, jitter, and packet loss, impacting voice first-class. it is critical to thoroughly configure Cisco QoS mechanisms end to conclusion throughout the community for correct audio and video performance.

    This chapter turned into excerpted with permissions from enforcing Cisco Unified Communications supervisor, half 2, with the aid of Chris Olsen, Copyright 2008. For extra information about this booklet and other titles, please visit Cisco Press.

    all through height periods, packets can't be despatched immediately on account of interface congestion. in its place, the packets are quickly stored in a queue, ready to be processed. The period of time the packet waits within the queue, known as the queuing prolong, can differ enormously in keeping with network situations and site visitors arrival costs. If the queue is full, newly obtained packets cannot be buffered anymore and get dropped (tail drop). figure 1-1 illustrates tail drop. Packets are processed on a primary in, first out (FIFO) mannequin in the hardware queue of all router interfaces. Voice conversations are predictable and relentless (sampling is every 20 milliseconds through default), however records functions are bursty and grasping. Voice for this reason is field to degradation of excellent on account of delay, jitter, and packet loss.

    determine 1-1

    click on for a bigger photo

    Bandwidth challenges

    each site in a multisite deployment constantly is interconnected through an IP WAN, or on occasion via a metropolitan-area network (MAN) such as Metro Ethernet. Bandwidth on WAN hyperlinks is restricted and comparatively expensive. The purpose is to use the purchasable bandwidth as effectively as possible. unnecessary site visitors should be removed from the IP WAN links through content material filtering, firewalls, and entry control lists (ACL). IP WAN acceleration strategies for bandwidth optimization should be regarded as smartly. Any length of congestion may outcome in provider degradation except QoS is deployed during the community.

    Voice streams are steady and predictable for Cisco audio packets. usually, the G.729 codec is used throughout the WAN to optimal use bandwidth. As a evaluation, the G.711 audio codec requires sixty four kbps, whereas packetizing the G.711 voice pattern in an IP/UDP/RTP header each 20 ms requires sixteen kbps plus the Layer 2 header overhead.

    Voice is sampled each 20 ms, resulting in 50 packets per second (pps). The IP header is 20 bytes, whereas the UDP header is 8 bytes, and the RTP header is 12 bytes. The 40 bytes of header information should be converted to bits to figure out the packet expense of the overhead. because a byte has eight bits, 40 bytes * 8 bits in a byte = 320 bits. The 320 bits are sent 50 times per second according to the 20-ms price (1 millisecond is 1/a thousand of a second, and 20/1000 = .02). So:

    .02 * 50 = 1 second320 bits * 50 = sixteen,000 bits/sec, or 16 kbps

    Voice packets are benign compared to the bandwidth consumed via data functions. data purposes can fill the whole optimum transmission unit (MTU) of an Ethernet body (1518 bytes or 9216 bytes if jumbo Ethernet frames had been enabled). In assessment to information utility packets, voice packets are very small (60 bytes for G.729 and 200 bytes for G.711 with the default 20-ms sampling fee).

    In determine 1-2, a convention bridge has been deployed on the main website. No convention bridge exists on the far flung web page. If three IP telephones at a remote web page be a part of a convention, their RTP streams are despatched throughout the WAN to the conference bridge. The convention bridge, no matter if the use of software or hardware materials, mixes the acquired audio streams after which sends lower back three interesting unicast audio streams to the IP phones over the IP WAN. The convention bridge gets rid of the receiver's voice from his or her entertaining RTP circulation so that the person does not adventure echo as a result of the extend of traversing the WAN link and mixing RTP audio streams within the convention bridge.

    figure 1-2

    click for a bigger picture

    Centralized convention materials trigger bandwidth, extend, and capacity challenges in the voice community. each and every G.711 RTP circulate requires 80 kbps (plus the Layer 2 overhead), leading to 240 kbps of IP WAN bandwidth consumption with the aid of this voice convention. If the conference bridge had been no longer located on the other aspect of the IP WAN, this traffic would now not should traverse the WAN hyperlink, resulting in less lengthen and bandwidth consumption. If the remote website had a CUCM location configuration that resulted in calls with the G.729 codec returned to the main website, the utility conferencing substances of CUCM would now not be capable of mix the audio conversations. Hardware conferencing or hardware transcoder media components in a voice gateway are required to accommodate G.729 audio conferencing. native hardware conference elements would remove this want. All centrally found media supplies (song On grasp [MOH], annunciator, conference bridges, videoconferencing, and media termination facets) undergo an identical bandwidth, extend, and useful resource exhaustion challenges.

    Availability challenges

    When deploying CUCM in multisite environments, centralized CUCM-based services are accessed over the IP WAN. Affected services include the following:

  • Signaling in CUCM multisite deployments with centralized call processing: remote Cisco IP phones register with a centralized CUCM server. far off MGCP gateways are controlled by way of a centralized CUCM server that acts as an MGCP call agent.
  • Signaling in CUCM multisite deployments with dispensed call processing: In such environments, websites are connected via H.323 (non-gatekeeper-controlled, gatekeeper-managed, or H.225) or Session Initiation Protocol (SIP) trunks.
  • Media exchange: RTP streams between endpoints determined at different websites.
  • different features: These include Cisco IP phone Extensible Markup Language (XML) functions and access to purposes reminiscent of attendant console, CUCM Assistant, and others.
  • determine 1-three indicates a Unified Communications network by which the leading site is linked to a remote web site via a centralized name-processing ambiance. The main web site is also related to a faraway cluster through an intercluster trunk (ICT) representing a distributed name processing atmosphere. The mixture of both centralized and dispensed call processing represents a hybrid name-processing mannequin through which small sites use the CUCM elements of the leading website, but massive faraway places of work have their personal CUCM cluster. On the bottom left of figure 1-3 is a SIP trunk, customarily over a Metro Ethernet connection to an online Telephony carrier issuer (ITSP). The advantage of the SIP trunk is that the ITSP offers the gateways to the PSTN as an alternative of your providing gateways on the leading web site.

    figure 1-3

    click on for a larger graphic

    An IP WAN outage in determine 1-three will trigger an outage of call-processing capabilities for the remote website connected in a centralized fashion. The remote cluster will no longer suffer a call-processing outage, however the far off cluster are usually not in a position to dial the main website over the IP WAN throughout the outage. Mission-important voice purposes (voice mail, interactive voice response [IVR], and so on) observed at the leading site can be unavailable to any of the different sites all over the WAN outage.

    If the ITSP is the use of the identical links that allow IP WAN connectivity, all calls to and from the public switched phone network (PSTN) will even be unavailable.

    A deployment like the one shown in determine 1-3 is considered badly designed on account of the shortcoming of IP WAN and PSTN backup.

    Dial plan challenges

    In a multisite deployment, with a single or varied CUCM clusters, dial plan design requires the dignity of a few issues that do not exist in single-web page deployments:

  • Overlapping numbers: users observed at distinctive sites can have the equal directory numbers assigned. as a result of directory numbers always are exciting best within a web site, a multisite deployment requires a solution for overlapping numbers.
  • Nonconsecutive numbers: Contiguous tiers of numbers are vital to summarize name-routing suggestions, analogous to contiguous IP tackle tiers for route summarization. Such blocks will also be represented by way of one or a few entries in a call-routing desk, equivalent to route patterns, dial peer destination patterns, and voice translation rules, which maintain the routing table short and easy. If every endpoint requires its own entry within the call-routing table, the desk gets too massive, lots of memory is required, and lookups take extra time. for this reason, nonconsecutive numbers at any site are not most desirable for productive name routing.
  • Variable-length numbering: Some countries, such as the U.S. and Canada, have mounted-size numbering plans for PSTN numbers. Others, corresponding to Mexico and England, have variable-length numbering plans. an issue with variable-size numbers is that the comprehensive length of the number dialed will also be determined best by way of the CUCM route plan by expecting the interdigit timeout. looking ahead to the interdigit timeout, typical because the T.302 timer, adds to the submit-dial delay, which may annoy users.
  • Direct inward dialing (DID) ranges and E.164 addressing: When considering that integration with the PSTN, internally used directory numbers must be regarding external PSTN numbers (E.164 addressing). counting on the numbering plan (fixed or variable) and capabilities offered by means of the PSTN, the following solutions are normal:
  • each internal listing quantity pertains to a hard and fast-length PSTN quantity: during this case, every internal listing number has its own dedicated PSTN quantity. The listing number can, but doesn't should, fit the least-giant digits of the PSTN quantity. In international locations with a hard and fast numbering plan, such as the North American Numbering Plan (NANP), this usually capacity that the four-digit office codes are used as interior directory numbers. If these aren't pleasing, digits of workplace codes or administratively assigned web site codes may be added, leading to 5 or more digits getting used for internal listing numbers.

    an additional answer is to now not reuse any digits of the PSTN quantity but to readily map each and every internally used directory number to any PSTN quantity assigned to the enterprise. in this case, the internal and exterior numbers will not have anything in general. If the internally used listing number matches the least-big digits of its corresponding PSTN quantity, significant digits may also be set on the gateway or trunk. also, typical external cell quantity masks, transformation masks, or prefixes may also be configured. here is authentic because all interior directory numbers are modified to absolutely qualified PSTN numbers in the same manner. a different example is if the inner listing quantity is composed of parts of the PSTN number and administratively assigned digits similar to website codes plus PSTN station codes, or distinct degrees, similar to PSTN station codes 4100 to 4180 that map to listing numbers 1100 to 1180, or totally unbiased mappings of internal directory numbers to PSTN numbers. if that's the case, one or more translation guidelines ought to be used for incoming calls, and one or greater calling party transformation suggestions, transformation masks, external cell number masks, or prefixes have to be configured.

  • No DID aid in mounted-length numbering plans: To keep away from the requirement of one PSTN quantity per internal listing number when the usage of a set-size numbering plan, it is typical to disallow DID to an extension. instead, the PSTN trunk has a single quantity, and all PSTN calls routed to that quantity are sent to an attendant, auto-attendant, receptionist, or secretary. From there, the calls are transferred to the acceptable interior extension.
  • inside directory numbers are part of a variable-length quantity: In international locations with variable-size numbering plans, a customarily shorter "subscriber" number is assigned to the PSTN trunk, but the PSTN routes all calls starting with this quantity to the trunk. The caller can add digits to identify the extension. There is not any fastened variety of extra digits or complete digits. besides the fact that children, there is a maximum, usually 32 digits, which provides the liberty to opt for the length of listing numbers. This highest size can be less. as an instance, in E.164 the optimum number is 15 digits, now not including the nation code. A caller with ease provides the acceptable extension to the business's (short) PSTN quantity when inserting a call to a particular consumer. If handiest the brief PSTN number without an extension is dialed, the call is routed to an attendant inside the business. Residential PSTN numbers are usually longer and do not permit further digits to be added; the feature simply described is attainable most effective on trunks.
  • category of number (TON) in ISDN: The calling number (the automatic number Identification [ANI]) of calls being acquired from the PSTN may also be represented in other ways:

    - As a seven-digit subscriber quantity- As a ten-digit number, together with the area code- In overseas format with the country code in front of the area code

    To standardize the ANI for all calls, the layout this is used must be time-honored, and the quantity has to be transformed as a consequence.

  • Optimized call routing: Having an IP WAN between websites with PSTN access in any respect websites allows for PSTN toll bypass through sending calls between websites over the IP WAN in its place of the use of the PSTN. In such eventualities, the PSTN may still be used as a backup route best in case of WAN failure. a different answer, which extends the conception of toll skip and might potentially in the reduction of toll expenses, is to also use the IP WAN for PSTN calls. With tail-conclusion hop-off (TEHO), the IP WAN is used as a good deal as viable, and the gateway this is closest to the dialed PSTN destination is used for the PSTN breakout.
  • Overlapping and nonconsecutive numbers

    In determine 1-four, Cisco IP phones at the main web page use directory numbers 1001 to 1099, 2000 to 2157, and 2365 to 2999. at the far off web page, 1001 to 1099 and 2158 to 2364 are used. These directory numbers have two issues. First, 1001 to 1099 overlap; these directory numbers exist at both websites, so they don't seem to be exciting all the way through the comprehensive deployment. This explanations a problem: If a person within the far off site dialed handiest the 4 digits 1001, which mobile would ring? This situation of overlapping dial plans has to be addressed by using digit manipulation. additionally, the nonconsecutive use of the range 2000 to 2999 (with some duplicate numbers at the two sites) would require a big number of additional entries in call-routing tables because the tiers can infrequently be summarized by using one (or a number of) entries.

    determine 1-four

    click on for a bigger picture

    fastened versus variable-length numbering plans

    a set numbering plan facets fastened-length enviornment codes and native numbers. An open numbering plan facets variance in size of area code or native number, or both, within the country.

    desk 1-1 contrasts the NANP and a variable-length numbering plan -- Germany's numbering plan during this example.

    table 1-1

    component Description fastened Numbering Plan (NANP) Variable-length Numbering Plan (Germany) country code A code of 1 to 3 digits is used to reach the particular telephone gadget for each nation or particular provider. attain the E.164 common from to peer all international country codes. 1 forty nine area code Used within many countries to route calls to a selected city, area, or particular carrier. depending on the nation or region, it will possibly also be known as a numbering plan enviornment, subscriber trunk dialing code, country wide destination code, or routing code. Three digits Three to 5 digits Subscriber quantity Represents the particular cell quantity to be dialed, nonetheless it doesn't encompass the nation code, area code (if relevant), foreign prefix, or trunk prefix. Three-digit trade code plus a four-digit station code Three or greater digits Trunk prefix The preliminary digits to be dialed in a domestic call, before the area code and the subscriber number. 1 0 access code a number it really is historically dialed first "to get out to the PSTN," utilized in PBXs and VoIP techniques. nine0 foreign prefix The code dialed earlier than an international number (country code, area code if any, after which subscriber quantity). 011 00 or + (+ is used by using cell phones)


  • inside the U.S.: 9-1-408-555-1234 or 1-555-1234 (inside the identical enviornment code)
  • U.S. to Germany: 9-011-49-404-132670
  • within Germany: 0-0-404-132670 or 0-132670 (in the identical enviornment code)
  • Germany to the U.S.: 0-00-1-408-555-1234 (word: the 1 in 00-1-408 is the U.S. country code, no longer the trunk prefix.)
  • The NANP PSTN number is 408-555-1234, DID isn't used, and all calls positioned to the leading web site are handled by way of an attendant. there's a far flung website in Germany with the E.164 PSTN number +forty nine 404 13267. 4-digit extensions are used on the German area, and DID is allowed as a result of digits can be delivered to the PSTN quantity. When calling the German workplace attendant (no longer figuring out a selected extension), U.S. users would dial 9-011-49-404-13267. word how the + is changed with the aid of the international prefix 011 and the access code 9. If the cellphone with extension 1001 may still be referred to as at once, 9-011-forty nine-404-13267-1001 needs to be dialed.

    Variable-size numbering, E.164 addressing and DID

    determine 1-5 illustrates an illustration by which the main web page with CUCM resides within the U.S. and a remote web site devoid of CUCM resides in Germany. The NANP PSTN quantity in the U.S. is 408-555-1234. be aware that DID is not used, as a result of all calls placed to the leading web page are handled by an attendant. A remote website in Germany has PSTN number +49 404 13267. 4-digit extensions are used at the German place, and DID is allowed as a result of digits can be added to the PSTN number. When calling the German office attendant (now not understanding a selected extension), U.S. clients would dial 9-011-forty nine-404-13267. If the cellphone with extension 1001 should be called at once, 9-011-forty nine-404-13267-1001 has to be dialed.

    determine 1-5

    click on for a larger photo

    The common sense of routing calls by CUCM over the WAN or through the PSTN is appropriately transparent to the mobilephone consumer.

    Optimized call routing and PSTN backup

    There are two methods to shop costs for PSTN calls in a multisite deployment:

  • complete pass: Calls between sites inside an organization that use the IP WAN as an alternative of the PSTN. The PSTN is used for intersite calls only if calls over the IP WAN aren't viable—either because of a WAN failure or because the name isn't admitted through call Admission control (CAC).
  • Tail-conclusion hop-off (TEHO): Extends the theory of toll pass by means of also the usage of the IP WAN for calls to the far flung locations within the PSTN. With TEHO, the IP WAN is used as a lot as possible, and PSTN breakout happens at the gateway that is found closest to the dialed PSTN destination. local PSTN breakout is used as a backup in case of IP WAN or CAC.
  • within the instance proven in determine 1-6, a call from Chicago to San Jose could be routed as follows:

  • The Chicago CUCM categorical consumer dials 9-1-408-555-6666, a PSTN telephone located in San Jose.
  • The name is routed from Chicago CUCM categorical Router to the San Jose CUCM cluster over the IP WAN with both SIP or H.323.
  • The San Jose CUCM routes the call to the San Jose gateway, which breaks out to the PSTN with what now becomes a native low in cost call to the San Jose PSTN.
  • The San Jose PSTN relevant workplace routes the call, and the mobilephone rings.
  • determine 1-6

    click for a bigger graphic

    If the WAN had been unavailable for any intent earlier than the name, the Chicago Gateway would should be appropriately configured to route the name with the appropriate digit manipulation throughout the PSTN at a doubtlessly bigger toll charge to the San Jose PSTN phone.

    NAT and security issues

    In single-website deployments, CUCM servers and IP telephones always use deepest IP addresses because there isn't any deserve to talk with the backyard IP world. NAT is not configured for the cell subnets, and assaults from the backyard are impossible. In multisite deployments, besides the fact that children, IP security (IPsec) digital private community (VPN) tunnels can be used between websites. The VPN tunnels enable most effective intersite verbal exchange; entry to the covered internal networks isn't feasible from the backyard—handiest from the other website through the tunnel. hence, attacks from the backyard are blocked on the gateway. To configure IPsec VPNs, the VPN tunnel have to be configured to terminate on both gateways in the diverse sites. occasionally here is no longer feasible; for instance, the two websites could be under diverse administration, or most likely safety guidelines do not permit the configuration of IPsec VPNs.

    In such a case, or when connecting to a public service reminiscent of an ITSP, NAT must be configured for CUCM servers and IP phones. Cisco calls this Hosted NAT Traversal for Session Border Controllers.

    In determine 1-7, business A and enterprise B each use IP community internally. To speak over the web, the inner most addresses are translated into public IP addresses. enterprise A makes use of public IP community A, and business B uses public IP community B. All CUCM servers and IP telephones can be reached from the cyber web and speak with every different.

    As quickly as CUCM servers and IP telephones may also be reached with public IP addresses, they're subject to assaults from the outdoor world, introducing competencies safety issues.

    figure 1-7

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