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300-101 exam Dumps Source : Download 100% Free 300-101 Dumps PDF
Test Code : 300-101
Test designation : Implementing Cisco IP Routing (ROUTE v2.0)
Vendor designation : Cisco
braindumps : 750 existent Questions
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Chapter 1, picking concerns in a Multisite Deployment
Deploying Cisco Unified Communications manager between dissimilar sites requires an acceptable dial plan, acceptable bandwidth, a top character of carrier (QoS) implementation and a design that may live on IP WAN failures.
This chapter will clarify the concerns that can arise in a multisite Cisco UC manager deployment, and insinuate specific options.
picking out concerns in a Multisite DeploymentTable of contents:1. Multisite deployment problem overview2. first-class challenges3. Bandwidth challenges4. Availability challenges5. Dial route challenges6. Overlapping and nonconsecutive numbers7. fastened versus variable-length numbering plans8. Variable-size numbering, E.164 addressing and DID9. Optimized designation routing and PSTN backup10. NAT and protection issues
Multisite deployment problem overview
previously known as Cisco call manager (CCM), Cisco Unified Communications supervisor (CUCM) multi-web site deployment challenges can encompass the following:
IP networks had been now not originally designed to raise precise-time site visitors; in its place, they had been designed for resiliency and frailty tolerance. each packet is processed one after the other in an IP community, every so often inflicting distinctive packets in a communications circulation to rob distinct paths to the destination. The distinctive paths in the community may too fill a special amount of packet loss, delay, and lengthen version (jitter) as a result of bandwidth, distance, and congestion differences. The vacation spot must exist capable of salvage hold of packets out of order and resequence these packets. This challenge is solved by the utilize of precise-Time Transport Protocol (RTP) sequence numbers and site visitors resequencing. When possible, it's foremost to now not rely completely on these RTP mechanisms. suitable community design, the usage of Cisco router Cisco specific Forwarding (CEF) switch cache know-how, performs per-vacation spot load sharing by means of default. Per-destination load sharing isn't an exemplar load-balancing paradigm, however ensures that each and every IP flow (voice call) takes the identical course.
Bandwidth is shared by varied clients and purposes, whereas the amount of bandwidth required for a person IP movement varies enormously impeccable through brief lapses of time. Most statistics purposes are very bursty, whereas Cisco true-time audio communications with RTP utilize the identical continual-bandwidth movement. The bandwidth purchasable for any application, together with CUCM and voice-bearer traffic, is unpredictable. birthright through top intervals, packets exigency to exist buffered in queues waiting to exist processed on account of network congestion. Queuing is a time era that any person who has ever skilled air flight is widely wide-spread with. in case you arrive at the airport, you must salvage in a line (queue), since the variety of ticket agents (bandwidth) available to determine you in is below the flow of site visitors arriving at the ticket counters (incoming IP traffic). If congestion occurs for too long, the queue (packet buffers) gets crammed up, and passengers are irritated (packets are dropped). bigger queuing delays and packet drops are greater doubtless on enormously loaded, slow-speed links reminiscent of WAN hyperlinks used between sites in a multisite environment. first-rate challenges are criterion on these sorts of hyperlinks, and you exigency to deal with them by imposing QoS. with out the utilize of QoS, voice packets adventure extend, jitter, and packet loss, impacting voice high-quality. it is censorious to correctly configure Cisco QoS mechanisms conclusion to finish birthright through the community for apropos audio and video performance.This chapter became excerpted with permissions from imposing Cisco Unified Communications manager, fraction 2, by means of Chris Olsen, Copyright 2008. For extra assistance about this publication and other titles, tickle debate with Cisco Press.
all the route through peak intervals, packets can't exist despatched automatically on account of interface congestion. as a substitute, the packets are quickly stored in a queue, waiting to exist processed. The amount of time the packet waits within the queue, called the queuing delay, can disagree tremendously in response to community situations and traffic arrival rates. If the queue is full, newly received packets cannot exist buffered anymore and salvage dropped (tail drop). device 1-1 illustrates tail drop. Packets are processed on a first in, first out (FIFO) mannequin in the hardware queue of impeccable router interfaces. Voice conversations are predictable and incessant (sampling is each 20 milliseconds via default), but records purposes are bursty and grasping. Voice for this understanding is topic to degradation of best on account of lengthen, jitter, and packet loss.
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each and every web site in a multisite deployment continually is interconnected by using an IP WAN, or on occasion by route of a metropolitan-enviornment network (MAN) comparable to Metro Ethernet. Bandwidth on WAN hyperlinks is restricted and comparatively expensive. The goal is to fabricate utilize of the obtainable bandwidth as correctly as possible. useless traffic should quiet exist removed from the IP WAN hyperlinks via content material filtering, firewalls, and access manage lists (ACL). IP WAN acceleration methods for bandwidth optimization should quiet exist regarded as well. Any duration of congestion might outcome in carrier degradation until QoS is deployed impeccable the route through the community.
Voice streams are constant and predictable for Cisco audio packets. usually, the G.729 codec is used across the WAN to optimum utilize bandwidth. As a assessment, the G.711 audio codec requires 64 kbps, whereas packetizing the G.711 voice pattern in an IP/UDP/RTP header every 20 ms requires 16 kbps plus the Layer 2 header overhead.
Voice is sampled each 20 ms, leading to 50 packets per second (pps). The IP header is 20 bytes, whereas the UDP header is eight bytes, and the RTP header is 12 bytes. The 40 bytes of header assistance ought to exist converted to bits to device out the packet fee of the overhead. because a byte has 8 bits, forty bytes * eight bits in a byte = 320 bits. The 320 bits are despatched 50 times per second based on the 20-ms expense (1 millisecond is 1/a thousand of a 2nd, 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 by route of records purposes. facts functions can fill the total highest transmission unit (MTU) of an Ethernet frame (1518 bytes or 9216 bytes if jumbo Ethernet frames fill been enabled). In evaluation to records software packets, voice packets are very minute (60 bytes for G.729 and 200 bytes for G.711 with the default 20-ms sampling price).
In device 1-2, a conference bridge has been deployed at the leading web site. No conference bridge exists on the far off web page. If three IP telephones at a far flung site exist a fraction of a convention, their RTP streams are sent across the WAN to the convention bridge. The conference bridge, whether the usage of application or hardware resources, mixes the got audio streams and then sends back three wonderful unicast audio streams to the IP telephones over the IP WAN. The conference bridge eliminates the receiver's voice from his or her challenging RTP circulation so that the consumer does not undergo reecho as a result of the extend of traversing the WAN hyperlink and mixing RTP audio streams within the convention bridge.
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Centralized convention supplies trigger bandwidth, lengthen, and means challenges in the voice network. each and every G.711 RTP flow requires 80 kbps (plus the Layer 2 overhead), resulting in 240 kbps of IP WAN bandwidth consumption by means of this voice convention. If the conference bridge were not create on the other facet of the IP WAN, this traffic would no longer deserve to traverse the WAN hyperlink, leading to much less prolong and bandwidth consumption. If the remote web site had a CUCM dwelling configuration that resulted in calls with the G.729 codec again to the main website, the software conferencing resources of CUCM would not exist able to combine the audio conversations. Hardware conferencing or hardware transcoder media supplies in a voice gateway are required to accommodate G.729 audio conferencing. local hardware convention supplies would salvage rid of this need. impeccable centrally discovered media resources (tune On hang [MOH], annunciator, convention bridges, videoconferencing, and media termination aspects) endure identical bandwidth, delay, and useful resource exhaustion challenges.
When deploying CUCM in multisite environments, centralized CUCM-based mostly functions are accessed over the IP WAN. Affected functions consist of here:
determine 1-3 indicates a Unified Communications community in which the main web page is linked to a far off web page through a centralized call-processing atmosphere. The main web site is additionally connected to a remote cluster through an intercluster trunk (ICT) representing a disbursed call processing ambiance. The aggregate of both centralized and disbursed call processing represents a hybrid name-processing mannequin during which minute websites utilize the CUCM supplies of the leading web page, but great far off workplaces fill their personal CUCM cluster. On the bottom left of determine 1-three is a SIP trunk, typically over a Metro Ethernet connection to an online Telephony carrier provider (ITSP). The advantage of the SIP trunk is that the ITSP gives the gateways to the PSTN as an alternative of your presenting gateways at the leading site.
click for a larger graphic
An IP WAN outage in device 1-three will understanding an outage of call-processing services for the far off web page related in a centralized trend. The remote cluster will not endure a name-processing outage, however the remote cluster aren't able to dial the main website over the IP WAN impeccable over the outage. Mission-critical voice applications (voice mail, interactive voice response [IVR], and the like) determined at the main web page could exist unavailable to any of the other sites impeccable the route through the WAN outage.
If the ITSP is the utilize of the identical hyperlinks that enable IP WAN connectivity, impeccable calls to and from the public switched telephone community (PSTN) will too exist unavailable.
A deployment devotion the one shown in determine 1-three is regarded badly designed as a result of the shortcoming of IP WAN and PSTN backup.
Dial route challenges
In a multisite deployment, with a separate or distinctive CUCM clusters, dial route design requires the distinction of a number of concerns that achieve not exist in single-website deployments:
each and every interior directory number pertains to a set-size PSTN number: in this case, each interior listing number has its personal dedicated PSTN number. The listing number can, but doesn't should, suit the least-colossal digits of the PSTN number. In nations with a fixed numbering plan, such as the North American Numbering route (NANP), this usually skill that the 4-digit office codes are used as interior listing numbers. If these are not enjoyable, digits of office codes or administratively assigned web site codes could exist delivered, resulting in five or extra digits getting used for inside listing numbers.
one other respond is to now not reuse any digits of the PSTN quantity however to effectively map each internally used listing quantity to any PSTN quantity assigned to the enterprise. in this case, the inside and external numbers achieve not exigency anything else in typical. If the internally used listing number matches the least-significant digits of its corresponding PSTN quantity, huge digits may too exist set at the gateway or trunk. additionally, commonplace exterior cell quantity masks, transformation masks, or prefixes can too exist configured. here is proper as a result of impeccable inner directory numbers are changed to totally certified PSTN numbers within the equal means. a different instance is if the internal directory quantity consists of constituents of the PSTN number and administratively assigned digits similar to site codes plus PSTN station codes, or distinctive ranges, such as PSTN station codes 4100 to 4180 that map to listing numbers 1100 to 1180, or absolutely independent mappings of inside listing numbers to PSTN numbers. in that case, one or more translation guidelines fill to exist used for incoming calls, and one or greater calling birthday party transformation rules, transformation masks, external mobile quantity masks, or prefixes ought to exist configured.
type of quantity (TON) in ISDN: The calling quantity (the computerized number Identification [ANI]) of calls being bought from the PSTN will too exist represented in alternative ways:
- As a seven-digit subscriber number- As a ten-digit quantity, together with the district code- In overseas format with the country code in front of the district code
To standardize the ANI for impeccable calls, the layout that's used fill to exist favourite, and the quantity must exist transformed hence.
Overlapping and nonconsecutive numbers
In determine 1-four, Cisco IP phones at the leading website utilize listing numbers 1001 to 1099, 2000 to 2157, and 2365 to 2999. at the faraway web page, 1001 to 1099 and 2158 to 2364 are used. These listing numbers fill two concerns. First, 1001 to 1099 overlap; these listing numbers exist at each websites, so that they aren't exciting throughout the complete deployment. This reasons an issue: If a person in the remote website dialed handiest the 4 digits 1001, which mobile would ring? This situation of overlapping dial plans must exist addressed by digit manipulation. moreover, the nonconsecutive utilize of the ambit 2000 to 2999 (with some duplicate numbers on the two websites) would require a major number of extra entries in name-routing tables since the stages can hardly exist summarized by means of one (or a few) entries.
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fixed versus variable-length numbering plans
a fixed numbering route elements fastened-size district codes and autochthonous numbers. An open numbering route facets variance in length of district code or autochthonous quantity, or both, within the country.
desk 1-1 contrasts the NANP and a variable-size numbering route -- Germany's numbering route during this illustration.
desk 1-1component Description fixed Numbering route (NANP) Variable-length Numbering route (Germany) country code A code of 1 to three digits is used to attain the particular cell system for each nation or special carrier. obtain the E.164 commonplace from http://itu.org to peer impeccable overseas country codes. 1 forty nine area code Used inside many countries to route calls to a selected city, area, or particular service. reckoning on the nation or region, it may too exist called a numbering route area, subscriber trunk dialing code, country wide destination code, or routing code. Three digits Three to 5 digits Subscriber quantity Represents the selected mobile quantity to exist dialed, nevertheless it doesn't consist of the nation code, enviornment code (if applicable), exotic prefix, or trunk prefix. Three-digit exchange code plus a four-digit station code Three or greater digits Trunk prefix The initial digits to exist dialed in a home name, before the enviornment code and the subscriber number. 1 0 entry code a host it really is traditionally dialed first "to salvage out to the PSTN," utilized in PBXs and VoIP methods. 90 international prefix The code dialed before an international quantity (nation code, district code if any, after which subscriber number). 011 00 or + (+ is used via cell phones)
The NANP PSTN number is 408-555-1234, DID is not used, and impeccable calls positioned to the main site are handled with the aid of an attendant. there's a faraway site in Germany with the E.164 PSTN number +49 404 13267. four-digit extensions are used on the German area, and DID is allowed because digits can exist introduced to the PSTN quantity. When calling the German office attendant (not realizing a selected extension), U.S. clients would dial 9-011-forty nine-404-13267. notice how the + is changed by the overseas prefix 011 and the access code 9. If the cell with extension 1001 may quiet exist referred to as without delay, 9-011-49-404-13267-1001 must exist dialed.
Variable-size numbering, E.164 addressing and DID
determine 1-5 illustrates an instance through which the main web page with CUCM resides within the U.S. and a far flung site with out CUCM resides in Germany. The NANP PSTN number in the U.S. is 408-555-1234. note that DID isn't used, as a result of impeccable calls positioned to the leading web page are handled with the aid of an attendant. A far flung site in Germany has PSTN number +49 404 13267. four-digit extensions are used at the German area, and DID is allowed as a result of digits will too exist added to the PSTN number. When calling the German workplace attendant (now not knowing a particular extension), U.S. clients would dial 9-011-49-404-13267. If the cellphone with extension 1001 should exist referred to as without delay, 9-011-49-404-13267-1001 has to exist dialed.
click for a larger graphic
The well-behaved judgment of routing calls by means of CUCM over the WAN or in the course of the PSTN is accurately transparent to the mobilephone user.
Optimized call routing and PSTN backup
There are two how to store expenses for PSTN calls in a multisite deployment:
within the instance proven in determine 1-6, a designation from Chicago to San Jose would exist routed as follows:
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If the WAN were unavailable for any intent before the name, the Chicago Gateway would exigency to exist effectively configured to route the call with the confiscate digit manipulation during the PSTN at a doubtlessly bigger toll charge to the San Jose PSTN telephone.
NAT and security issues
In single-site deployments, CUCM servers and IP phones constantly utilize inner most IP addresses as a result of there isn't any should talk with the outside IP world. NAT isn't configured for the phone subnets, and attacks from the outside are not possible. In multisite deployments, although, IP protection (IPsec) digital deepest network (VPN) tunnels will too exist used between sites. The VPN tunnels permit only intersite verbal exchange; access to the included inside networks is not feasible from the outside—best from the different web page during the tunnel. therefore, attacks from the backyard are blocked at the gateway. To configure IPsec VPNs, the VPN tunnel ought to exist configured to terminate on both gateways within the diverse websites. on occasion this is now not viable; for example, both websites may exist under distinctive administration, or perhaps security guidelines don't allow the configuration of IPsec VPNs.
In one of these case, or when connecting to a public carrier similar to an ITSP, NAT must exist configured for CUCM servers and IP phones. Cisco calls this Hosted NAT Traversal for Session verge Controllers.
In device 1-7, enterprise A and company B each utilize IP community 10.0.0.0/8 internally. To discourse over the information superhighway, the deepest addresses are translated into public IP addresses. company A uses public IP network A, and company B uses public IP community B. impeccable CUCM servers and IP telephones may too exist reached from the information superhighway and talk with each and every different.
As soon as CUCM servers and IP phones may too exist reached with public IP addresses, they are discipline to attacks from the outdoor world, introducing competencies protection concerns.
click for a bigger photoThis down load first appeared at searchunifiedcommunications.com
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