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Test Number : 300-115
Test Name : Implementing Cisco IP Switched Networks (SWITCH v2.0)
Vendor Name : Cisco
: 523 Dumps Questions
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Chapter 1, making a choice on issues in a Multisite Deployment
Deploying Cisco Unified Communications supervisor between multiple sites requires an appropriate dial plan, acceptable bandwidth, a high quality of service (QoS) implementation and a design that can continue to exist IP WAN screw ups.
This chapter will explain the considerations that can arise in a multisite Cisco UC supervisor deployment, and recommend certain solutions.
opting for issues in a Multisite DeploymentTable of contents:1. Multisite deployment problem overview2. first-rate 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 issues
Multisite deployment challenge overview
formerly called Cisco call manager (CCM), Cisco Unified Communications manager (CUCM) multi-website deployment challenges can encompass the following:
IP networks were not at the beginning designed to raise actual-time traffic; in its place, they have been designed for resiliency and fault tolerance. each and every packet is processed one by one in an IP network, on occasion causing distinctive packets in a communications stream to take different paths to the destination. The different paths within the community may additionally have a special amount of packet loss, prolong, and delay variation (jitter) because of bandwidth, distance, and congestion transformations. The vacation spot should be capable of receive packets out of order and resequence these packets. This problem is solved by way of true-Time Transport Protocol (RTP) sequence numbers and site visitors resequencing. When viable, it's premier to no longer depend entirely on these RTP mechanisms. suitable community design, using Cisco router Cisco categorical Forwarding (CEF) switch cache expertise, performs per-vacation spot load sharing by default. Per-vacation spot load sharing isn't an ideal load-balancing paradigm, nonetheless it ensures that each and every IP movement (voice name) takes the identical course.
Bandwidth is shared via numerous users and functions, whereas the amount of bandwidth required for an individual IP move varies enormously all over short lapses of time. Most information functions are very bursty, whereas Cisco actual-time audio communications with RTP use the equal continuous-bandwidth stream. The bandwidth purchasable for any software, together with CUCM and voice-bearer traffic, is unpredictable. during top intervals, packets should be buffered in queues waiting to be processed as a result of network congestion. Queuing is a time period that anybody who has ever experienced air flight is time-honored with. if you happen to arrive on the airport, you should get in a line (queue), since the variety of ticket agents (bandwidth) obtainable to investigate you in is less than the circulate of traffic arriving at the ticket counters (incoming IP traffic). If congestion happens for too lengthy, the queue (packet buffers) receives filled up, and passengers are annoyed (packets are dropped). better queuing delays and packet drops are greater probably on highly loaded, gradual-velocity links comparable to WAN hyperlinks used between sites in a multisite atmosphere. excellent challenges are common on these styles of links, and also you deserve to address them by means of imposing QoS. with out the use of QoS, voice packets journey prolong, jitter, and packet loss, impacting voice best. it is important to effectively configure Cisco QoS mechanisms conclusion to end throughout the network for relevant audio and video efficiency.This chapter changed into excerpted with permissions from imposing Cisco Unified Communications manager, half 2, by means of Chris Olsen, Copyright 2008. For extra suggestions about this publication and other titles, please discuss with Cisco Press.
all through top intervals, packets can not be sent automatically on account of interface congestion. as a substitute, the packets are temporarily saved in a queue, ready to be processed. The amount of time the packet waits within the queue, known as the queuing extend, can differ enormously according to network situations and site visitors arrival prices. If the queue is full, newly received packets can't be buffered anymore and get dropped (tail drop). figure 1-1 illustrates tail drop. Packets are processed on a first in, first out (FIFO) mannequin within the hardware queue of all router interfaces. Voice conversations are predictable and relentless (sampling is every 20 milliseconds through default), but statistics purposes are bursty and greedy. Voice hence is area to degradation of first-class as a result of delay, jitter, and packet loss.
click for a bigger photo
each web page in a multisite deployment usually is interconnected by using an IP WAN, or from time to time through a metropolitan-enviornment network (MAN) comparable to Metro Ethernet. Bandwidth on WAN links is limited and relatively costly. The aim is to make use of the attainable bandwidth as efficaciously as viable. pointless site visitors should still be faraway from the IP WAN links via content material filtering, firewalls, and access control lists (ACL). IP WAN acceleration methods for bandwidth optimization should still be considered as smartly. Any period of congestion could outcome in carrier degradation until QoS is deployed during the community.
Voice streams are consistent and predictable for Cisco audio packets. customarily, the G.729 codec is used throughout the WAN to most reliable use bandwidth. As a evaluation, the G.711 audio codec requires 64 kbps, whereas packetizing the G.711 voice sample 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 forty bytes of header tips need to be converted to bits to work out the packet fee of the overhead. as a result of a byte has 8 bits, 40 bytes * eight bits in a byte = 320 bits. The 320 bits are despatched 50 times per 2d in keeping with the 20-ms fee (1 millisecond is 1/one thousand of a 2d, and 20/a thousand = .02). So:
.02 * 50 = 1 second320 bits * 50 = 16,000 bits/sec, or sixteen kbps
Voice packets are benign in comparison to the bandwidth consumed via information purposes. records purposes can fill the whole maximum transmission unit (MTU) of an Ethernet frame (1518 bytes or 9216 bytes if jumbo Ethernet frames were enabled). In evaluation to facts application packets, voice packets are very small (60 bytes for G.729 and 200 bytes for G.711 with the default 20-ms sampling rate).
In figure 1-2, a conference bridge has been deployed at the leading web site. No conference bridge exists on the far flung site. If three IP telephones at a remote web page be a part of a conference, their RTP streams are sent across the WAN to the convention bridge. The convention bridge, even if the use of application or hardware materials, mixes the obtained audio streams after which sends lower back three enjoyable unicast audio streams to the IP phones over the IP WAN. The conference bridge gets rid of the receiver's voice from his or her entertaining RTP movement in order that the user does not adventure echo because of the extend of traversing the WAN link and mixing RTP audio streams within the convention bridge.
click for a bigger graphic
Centralized convention substances cause bandwidth, prolong, and skill challenges within the voice network. every G.711 RTP movement requires 80 kbps (plus the Layer 2 overhead), leading to 240 kbps of IP WAN bandwidth consumption by way of this voice conference. If the convention bridge were no longer determined on the different facet of the IP WAN, this traffic would not need to traverse the WAN link, leading to less lengthen and bandwidth consumption. If the far off website had a CUCM location configuration that resulted in calls with the G.729 codec again to the leading web page, the software conferencing substances of CUCM would no longer be able to combine the audio conversations. Hardware conferencing or hardware transcoder media elements in a voice gateway are required to accommodate G.729 audio conferencing. local hardware conference materials would eliminate this need. All centrally discovered media components (tune On dangle [MOH], annunciator, convention bridges, videoconferencing, and media termination facets) undergo an identical bandwidth, lengthen, and resource exhaustion challenges.
When deploying CUCM in multisite environments, centralized CUCM-based mostly features are accessed over the IP WAN. Affected features encompass here:
figure 1-three suggests a Unified Communications community through which the main website is linked to a far off web page via a centralized call-processing atmosphere. The main web page is additionally connected to a far flung cluster through an intercluster trunk (ICT) representing a distributed name processing atmosphere. The mixture of both centralized and disbursed call processing represents a hybrid call-processing mannequin by which small websites use the CUCM materials of the leading web site, but big far off offices have their personal CUCM cluster. On the bottom left of figure 1-three is a SIP trunk, usually over a Metro Ethernet connection to an online Telephony carrier company (ITSP). The benefit of the SIP trunk is that the ITSP provides the gateways to the PSTN in its place of your featuring gateways at the main web site.
click for a larger image
An IP WAN outage in determine 1-three will cause an outage of call-processing functions for the far flung web page related in a centralized trend. The far off cluster will not suffer a name-processing outage, however the far flung cluster aren't capable of dial the leading web page over the IP WAN throughout the outage. Mission-vital voice purposes (voice mail, interactive voice response [IVR], etc) discovered at the leading site could be unavailable to any of the different sites right through the WAN outage.
If the ITSP is using the equal links that allow IP WAN connectivity, all calls to and from the public switched phone community (PSTN) will also be unavailable.
A deployment like the one shown in figure 1-3 is considered badly designed as a result of the lack of IP WAN and PSTN backup.
Dial plan challenges
In a multisite deployment, with a single or diverse CUCM clusters, dial plan design requires the honour of a couple of considerations that do not exist in single-website deployments:
each and every inside listing number pertains to a set-length PSTN quantity: during this case, each internal directory number has its own committed PSTN number. The directory quantity can, but does not need to, healthy the least-gigantic digits of the PSTN number. In nations with a fixed numbering plan, such because the North American Numbering Plan (NANP), this usually ability that the 4-digit office codes are used as inside directory numbers. If these aren't interesting, digits of office codes or administratively assigned web page codes might possibly be introduced, resulting in five or extra digits being used for inside listing numbers.
yet another answer is to not reuse any digits of the PSTN number however to readily map each and every internally used directory number to any PSTN number assigned to the company. during this case, the internal and exterior numbers won't have the rest in general. If the internally used listing number suits the least-giant digits of its corresponding PSTN quantity, gigantic digits may also be set at the gateway or trunk. additionally, normal exterior phone quantity masks, transformation masks, or prefixes can also be configured. here is genuine as a result of all inner directory numbers are changed to wholly certified PSTN numbers within the equal approach. one other illustration is that if the inner directory number consists of materials of the PSTN quantity and administratively assigned digits akin to web site codes plus PSTN station codes, or distinct stages, akin to PSTN station codes 4100 to 4180 that map to directory numbers 1100 to 1180, or absolutely independent mappings of internal directory numbers to PSTN numbers. if that's the case, one or extra translation rules have to be used for incoming calls, and one or greater calling party transformation suggestions, transformation masks, exterior phone quantity masks, or prefixes have to be configured.
classification of number (TON) in ISDN: The calling quantity (the automated quantity Identification [ANI]) of calls being obtained from the PSTN can be represented in other ways:
- As a seven-digit subscriber quantity- As a ten-digit number, together with the enviornment code- In international structure with the country code in entrance of the enviornment code
To standardize the ANI for all calls, the layout it really is used should be known, and the quantity has to be transformed hence.
Overlapping and nonconsecutive numbers
In figure 1-4, Cisco IP phones at the leading web site use directory numbers 1001 to 1099, 2000 to 2157, and 2365 to 2999. at the remote web site, 1001 to 1099 and 2158 to 2364 are used. These listing numbers have two considerations. First, 1001 to 1099 overlap; these directory numbers exist at each websites, so that they are not entertaining all through the finished deployment. This factors a problem: If a user within the faraway website dialed best the four digits 1001, which cell would ring? This difficulty of overlapping dial plans has to be addressed by using digit manipulation. furthermore, the nonconsecutive use of the latitude 2000 to 2999 (with some reproduction numbers at the two websites) would require a major variety of additional entries in call-routing tables since the ranges can hardly be summarized by way of one (or a number of) entries.
click on for a bigger photograph
fastened versus variable-length numbering plans
a fixed numbering plan elements mounted-length enviornment codes and local numbers. An open numbering plan points variance in length of enviornment code or local quantity, or both, inside the nation.
desk 1-1 contrasts the NANP and a variable-size numbering plan -- Germany's numbering plan in this illustration.
table 1-1part Description mounted Numbering Plan (NANP) Variable-length Numbering Plan (Germany) nation code A code of 1 to 3 digits is used to reach the selected mobile device for every nation or special service. attain the E.164 common from http://itu.org to look all overseas country codes. 1 forty nine enviornment code Used inside many countries to route calls to a selected metropolis, place, or special service. counting on the nation or place, it could even be referred to as a numbering plan area, subscriber trunk dialing code, country wide destination code, or routing code. Three digits Three to 5 digits Subscriber number Represents the particular telephone quantity to be dialed, nonetheless it does not consist of the nation code, area code (if relevant), overseas prefix, or trunk prefix. Three-digit exchange code plus a 4-digit station code Three or extra digits Trunk prefix The initial digits to be dialed in a home call, before the area code and the subscriber quantity. 1 0 entry code a bunch it really is historically dialed first "to get out to the PSTN," utilized in PBXs and VoIP programs. nine0 international prefix The code dialed before a world number (nation code, area code if any, after which subscriber number). 011 00 or + (+ is used with the aid of mobile phones)
The NANP PSTN number is 408-555-1234, DID isn't used, and all calls positioned to the main web site are dealt with with the aid of an attendant. there's a far off web site in Germany with the E.164 PSTN quantity +forty nine 404 13267. 4-digit extensions are used on the German vicinity, and DID is allowed as a result of digits can also be delivered to the PSTN quantity. When calling the German workplace attendant (now not realizing a particular extension), U.S. users would dial 9-011-49-404-13267. notice how the + is changed by the foreign prefix 011 and the access code 9. If the phone with extension 1001 may still be referred to as without delay, 9-011-forty nine-404-13267-1001 needs to be dialed.
Variable-length numbering, E.164 addressing and DID
determine 1-5 illustrates an instance in which the main web site with CUCM resides in 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 main website are dealt with by an attendant. A far flung website in Germany has PSTN quantity +49 404 13267. 4-digit extensions are used on the German vicinity, and DID is allowed as a result of digits can be introduced to the PSTN number. When calling the German office attendant (not figuring out a particular extension), U.S. clients would dial 9-011-49-404-13267. If the phone with extension 1001 may still be called directly, 9-011-forty nine-404-13267-1001 must be dialed.
click for a bigger picture
The good judgment of routing calls via CUCM over the WAN or during the PSTN is correctly clear to the cellphone user.
Optimized call routing and PSTN backup
There are two the way to keep costs for PSTN calls in a multisite deployment:
in the instance proven in figure 1-6, a call from Chicago to San Jose can be routed as follows:
click on for a larger image
If the WAN have been unavailable for any motive before the call, the Chicago Gateway would need to be accurately configured to route the name with the acceptable digit manipulation throughout the PSTN at a doubtlessly better toll can charge to the San Jose PSTN cell.
NAT and security considerations
In single-site deployments, CUCM servers and IP phones always use deepest IP addresses because there is no need to communicate with the backyard IP world. NAT isn't configured for the mobilephone subnets, and assaults from the backyard are unimaginable. In multisite deployments, however, IP safety (IPsec) digital private network (VPN) tunnels can also be used between websites. The VPN tunnels allow simplest intersite verbal exchange; entry to the covered interior networks isn't feasible from the outdoor—simplest from the other site in the course of the tunnel. therefore, attacks from the backyard are blocked on the gateway. To configure IPsec VPNs, the VPN tunnel must be configured to terminate on both gateways in the diverse sites. now and again here is now not possible; as an instance, the two sites can be beneath distinctive administration, or perhaps protection guidelines do not permit the configuration of IPsec VPNs.
In one of these case, or when connecting to a public provider corresponding to an ITSP, NAT has to be configured for CUCM servers and IP telephones. Cisco calls this Hosted NAT Traversal for Session Border Controllers.
In figure 1-7, enterprise A and business B both use IP community 10.0.0.0/eight internally. To talk over the web, the inner most addresses are translated into public IP addresses. company A uses public IP community A, and company B makes use of public IP network B. All CUCM servers and IP telephones can also be reached from the internet and speak with each and every different.
As quickly as CUCM servers and IP telephones can be reached with public IP addresses, they're field to assaults from the outside world, introducing skills safety concerns.
click on for a larger photographThis obtain first regarded at searchunifiedcommunications.com
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