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Test Code : 300-075
Test title : Implementing Cisco IP Telephony and Video, section 2 (CIPTV2)
Vendor title : Cisco
: 649 real Questions

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Implementing Cisco IP Telephony and Video, section 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 property 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 design challenges6. Overlapping and nonconsecutive numbers7. mounted versus variable-size numbering plans8. Variable-length numbering, E.164 addressing and DID9. Optimized title routing and PSTN backup10. NAT and safety considerations

Multisite deployment problem overview

previously known as Cisco convene 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 necessity to subsist prioritized over a packet-switching network. sum site visitors is treated equally by default in routers and switches. Voice and video are delay-sensitive packets that necessity to subsist 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 subsist calculated and provisioned for to construct certain that information purposes and Cisco Unified Communications applications don't overload the available bandwidth. Bandwidth reservations will besides subsist made to functions through QoS deployment.
  • Availability issues: When deploying Cisco Unified Communications manager (CUCM) with centralized title processing, IP telephones register with CUCM over the IP LAN and doubtlessly over the WAN. If gateways in far flung sites are the exercise of Media Gateway manage Protocol (MGCP) as a signaling protocol, they additionally depend upon the availability of CUCM appearing as an MGCP convene agent. it is captious to set aside in constrain 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 design considerations: directory numbers (DN) can overlap throughout numerous sites. Overlapping dial plans and nonconsecutive numbers may besides subsist solved via designing a tenacious multisite dial plan. prevent overlapping numbers across websites each time viable for a simpler design.
  • NAT and security concerns: the exercise of inner most IP addresses inside an trade 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 trade believe to subsist translated into public IP addresses. Public IP addresses construct the IP phones seen from the internet and therefore zone to assaults.
  • high-quality challenges

    IP networks were no longer at the genesis 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 bewitch diverse paths to the vacation spot. The different paths in the community may besides believe a divide amount of packet loss, prolong, and prolong version (jitter) on account of bandwidth, distance, and congestion modifications. The destination ought to subsist able to salvage 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. redress 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 faultless 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 flux varies vastly throughout short lapses of time. Most records purposes are very bursty, whereas Cisco precise-time audio communications with RTP exercise the equal continual-bandwidth movement. The bandwidth available for any software, together with CUCM and voice-bearer traffic, is unpredictable. sum the pass through height intervals, packets should subsist buffered in queues ready to subsist 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 necessity to salvage 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 workable 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 maneuver them by enforcing QoS. devoid of the exercise of QoS, voice packets savor lengthen, jitter, and packet loss, impacting voice first-class. it is captious to thoroughly configure Cisco QoS mechanisms discontinuance to conclusion throughout the community for redress 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, gratify visit Cisco Press.

    all through height periods, packets can't subsist despatched immediately on account of interface congestion. in its place, the packets are quickly stored in a queue, ready to subsist processed. The age of time the packet waits within the queue, known as the queuing prolong, can vary enormously in keeping with network situations and site visitors arrival costs. If the queue is full, newly obtained packets cannot subsist buffered anymore and salvage dropped (tail drop). motif 1-1 illustrates tail drop. Packets are processed on a primary in, first out (FIFO) mannequin in the hardware queue of sum 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 judgement is realm 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 exercise the purchasable bandwidth as effectively as possible. unnecessary site visitors should subsist removed from the IP WAN links through content material filtering, firewalls, and entry control lists (ACL). IP WAN acceleration strategies for bandwidth optimization should subsist regarded as smartly. Any length of congestion may outcome in provider degradation except QoS is deployed during the community.

    Voice streams are even and predictable for Cisco audio packets. usually, the G.729 codec is used throughout the WAN to optimal exercise 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 subsist converted to bits to motif 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 charge (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 all 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 diminutive (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 subsist a section of a convention, their RTP streams are despatched throughout the WAN to the conference bridge. The convention bridge, no matter if the exercise of software or hardware materials, mixes the acquired audio streams after which sends lower back three exciting 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 subsist capable of fuse the audio conversations. Hardware conferencing or hardware transcoder media components in a voice gateway are required to accommodate G.729 audio conferencing. indigenous hardware conference elements would remove this want. sum centrally institute 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 convene processing: remote Cisco IP phones register with a centralized CUCM server. far off MGCP gateways are controlled by pass of a centralized CUCM server that acts as an MGCP convene agent.
  • Signaling in CUCM multisite deployments with dispensed convene 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 besides related to a faraway cluster through an intercluster trunk (ICT) representing a distributed title processing atmosphere. The blend of both centralized and dispensed convene processing represents a hybrid name-processing mannequin through which diminutive sites exercise the CUCM elements of the leading website, but massive faraway places of labor believe their personal CUCM cluster. On the bottom left of motif 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 subsist unavailable to any of the different sites sum over the WAN outage.

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

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

    Dial design challenges

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

  • Overlapping numbers: users observed at distinctive sites can believe 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 besides subsist represented by pass 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 recollection is required, and lookups bewitch extra time. for this reason, nonconsecutive numbers at any site are not most desirable for productive title routing.
  • Variable-length numbering: Some countries, such as the U.S. and Canada, believe mounted-size numbering plans for PSTN numbers. Others, corresponding to Mexico and England, believe variable-length numbering plans. an issue with variable-size numbers is that the comprehensive length of the number dialed will besides subsist determined best by pass of the CUCM route design 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 subsist regarding external PSTN numbers (E.164 addressing). counting on the numbering design (fixed or variable) and capabilities offered by means of the PSTN, the following solutions are normal:
  • each internal listing quantity pertains to a difficult 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, meet the least-giant digits of the PSTN quantity. In international locations with a difficult and rapidly numbering plan, such as the North American Numbering design (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 subsist added, leading to 5 or more digits getting used for internal listing numbers.

    an additional own 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 believe anything in general. If the internally used listing number matches the least-big digits of its corresponding PSTN quantity, significant digits may besides subsist set on the gateway or trunk. also, typical external cell quantity masks, transformation masks, or prefixes may besides subsist configured. here is undoubted because sum interior directory numbers are modified to absolutely qualified PSTN numbers in the selfsame manner. a different instance 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 divide 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 subsist used for incoming calls, and one or greater calling party transformation suggestions, transformation masks, external cell number masks, or prefixes believe to subsist configured.

  • No DID aid in mounted-length numbering plans: To hold 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 sole quantity, and sum 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 section 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 sum 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 subsist 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 convene to a particular consumer. If handiest the brief PSTN number without an extension is dialed, the convene is routed to an attendant inside the business. Residential PSTN numbers are usually longer and attain not permit further digits to subsist 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 besides subsist represented in other ways:

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

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

  • Optimized convene 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 position of the exercise of the PSTN. In such eventualities, the PSTN may noiseless subsist 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 besides exercise the IP WAN for PSTN calls. With tail-conclusion hop-off (TEHO), the IP WAN is used as a sociable 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 exercise 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 believe two issues. First, 1001 to 1099 overlap; these directory numbers exist at both websites, so they don't look to subsist exciting sum the pass 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 subsist addressed by using digit manipulation. additionally, the nonconsecutive exercise of the compass 2000 to 2999 (with some duplicate numbers at the two sites) would require a tall number of additional entries in call-routing tables because the tiers can infrequently subsist 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 design facets fastened-length enviornment codes and indigenous numbers. An open numbering design facets variance in size of zone code or indigenous number, or both, within the country.

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

    table 1-1

    component Description fastened Numbering design (NANP) Variable-length Numbering design (Germany) country code A code of 1 to 3 digits is used to compass the particular telephone gadget for each nation or particular provider. attain the E.164 common from http://itu.org to peer sum 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 besides subsist known as a numbering design 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 subsist dialed, nonetheless it doesn't encompass the nation code, zone code (if relevant), alien prefix, or trunk prefix. Three-digit trade code plus a four-digit station code Three or greater digits Trunk prefix The prefatory digits to subsist dialed in a domestic call, before the zone code and the subscriber number. 1 0 access code a number it really is historically dialed first "to salvage out to the PSTN," utilized in PBXs and VoIP techniques. nine0 foreign prefix The code dialed earlier than an international number (country code, zone 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 sum calls positioned to the leading web site are handled by pass 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 subsist 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 noiseless subsist referred to as at once, 9-011-forty nine-404-13267-1001 needs to subsist 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. subsist vigilant that DID is not used, as a result of sum 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 subsist 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 subsist called at once, 9-011-forty nine-404-13267-1001 has to subsist 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 convene 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 exercise 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 title isn't admitted through convene Admission control (CAC).
  • Tail-conclusion hop-off (TEHO): Extends the theory of toll pass by means of besides 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 institute 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 convene from Chicago to San Jose could subsist routed as follows:

  • The Chicago CUCM categorical consumer dials 9-1-408-555-6666, a PSTN telephone located in San Jose.
  • The title 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 convene to the San Jose gateway, which breaks out to the PSTN with what now becomes a indigenous low in cost convene to the San Jose PSTN.
  • The San Jose PSTN apposite 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 subsist appropriately configured to route the title with the arrogate 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 exercise 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 subsist 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 believe to subsist configured to terminate on both gateways in the diverse sites. occasionally here is no longer feasible; for instance, the two websites could subsist under diverse administration, or most likely safety guidelines attain not permit the configuration of IPsec VPNs.

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

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

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

    figure 1-7

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    This down load first appeared at searchunifiedcommunications.com

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