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Test Number : 300-070
Test Name : Implementing Cisco IP Telephony and Video, Part 1 (CIPTV1)
Vendor Name : Cisco
: 211 Dumps Questions
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Chapter 1, deciding on concerns in a Multisite Deployment
Deploying Cisco Unified Communications supervisor between multiple sites requires a suitable dial plan, acceptable bandwidth, a top quality of carrier (QoS) implementation and a design that can survive IP WAN disasters.
This chapter will explain the considerations that may arise in a multisite Cisco UC manager deployment, and suggest certain options.
making a choice on issues in a Multisite DeploymentTable of contents:1. Multisite deployment challenge overview2. first-class challenges3. Bandwidth challenges4. Availability challenges5. Dial plan challenges6. Overlapping and nonconsecutive numbers7. fastened versus variable-size numbering plans8. Variable-size numbering, E.164 addressing and DID9. Optimized call routing and PSTN backup10. NAT and safety issues
Multisite deployment challenge overview
previously called Cisco call manager (CCM), Cisco Unified Communications manager (CUCM) multi-site deployment challenges can encompass here:
IP networks were not initially designed to raise precise-time site visitors; as an alternative, they had been designed for resiliency and fault tolerance. each packet is processed one by one in an IP network, now and again inflicting different packets in a communications stream to take distinct paths to the destination. The diverse paths within the community may also have a unique amount of packet loss, lengthen, and prolong version (jitter) because of bandwidth, distance, and congestion ameliorations. The destination must be in a position to obtain packets out of order and resequence these packets. This challenge is solved via true-Time Transport Protocol (RTP) sequence numbers and traffic resequencing. When viable, it's most desirable to no longer depend totally on these RTP mechanisms. correct community design, using Cisco router Cisco categorical Forwarding (CEF) switch cache technology, performs per-destination load sharing by means of default. Per-vacation spot load sharing isn't an ideal load-balancing paradigm, but it surely ensures that each IP flow (voice name) takes the identical route.
Bandwidth is shared through multiple users and purposes, whereas the amount of bandwidth required for an individual IP circulate varies enormously during short lapses of time. Most statistics purposes are very bursty, whereas Cisco actual-time audio communications with RTP use the identical continuous-bandwidth circulation. The bandwidth accessible for any software, including CUCM and voice-bearer traffic, is unpredictable. throughout top durations, packets deserve to be buffered in queues ready to be processed on account of community congestion. Queuing is a term that anybody who has ever experienced air flight is widespread with. in case you arrive on the airport, you have to get in a line (queue), because the variety of ticket agents (bandwidth) accessible to assess you in is less than the circulate of site visitors arriving on the ticket counters (incoming IP site visitors). If congestion occurs for too lengthy, the queue (packet buffers) gets stuffed up, and passengers are aggravated (packets are dropped). greater queuing delays and packet drops are more probably on highly loaded, sluggish-speed links equivalent to WAN hyperlinks used between sites in a multisite environment. excellent challenges are usual on these sorts of links, and also you deserve to address them with the aid of imposing QoS. devoid of using QoS, voice packets experience extend, jitter, and packet loss, impacting voice excellent. it is vital to properly configure Cisco QoS mechanisms conclusion to end during the network for proper audio and video performance.This chapter changed into excerpted with permissions from imposing Cisco Unified Communications manager, part 2, with the aid of Chris Olsen, Copyright 2008. For greater information about this e-book and different titles, please visit Cisco Press.
throughout height periods, packets can't be despatched instantly on account of interface congestion. as a substitute, the packets are briefly kept in a queue, waiting to be processed. The amount of time the packet waits in the queue, known as the queuing extend, can fluctuate tremendously in keeping with network circumstances and traffic arrival rates. If the queue is full, newly obtained 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 in the hardware queue of all router interfaces. Voice conversations are predictable and constant (sampling is every 20 milliseconds by means of default), but statistics applications are bursty and greedy. Voice hence is subject to degradation of excellent because of prolong, jitter, and packet loss.
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each and every site in a multisite deployment usually is interconnected by an IP WAN, or from time to time by using a metropolitan-area community (MAN) comparable to Metro Ethernet. Bandwidth on WAN hyperlinks is restricted and relatively high priced. The intention is to use the available bandwidth as correctly as feasible. unnecessary site visitors may still be removed from the IP WAN links via content filtering, firewalls, and access control lists (ACL). IP WAN acceleration methods for bandwidth optimization should still be regarded as neatly. Any period of congestion could result in carrier degradation unless QoS is deployed all the way through the community.
Voice streams are regular and predictable for Cisco audio packets. customarily, the G.729 codec is used across the WAN to surest 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 16 kbps plus the Layer 2 header overhead.
Voice is sampled every 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 information must be converted to bits to figure out the packet cost of the overhead. as a result of a byte has eight bits, 40 bytes * 8 bits in a byte = 320 bits. The 320 bits are despatched 50 times per 2nd in accordance with the 20-ms cost (1 millisecond is 1/a thousand of a second, and 20/a thousand = .02). So:
.02 * 50 = 1 second320 bits * 50 = 16,000 bits/sec, or 16 kbps
Voice packets are benign in comparison to the bandwidth consumed by means of data applications. records functions can fill the entire maximum transmission unit (MTU) of an Ethernet body (1518 bytes or 9216 bytes if jumbo Ethernet frames were enabled). In comparison to facts utility packets, voice packets are very small (60 bytes for G.729 and 200 bytes for G.711 with the default 20-ms sampling cost).
In figure 1-2, a convention bridge has been deployed at the leading website. No conference bridge exists at the far flung site. If three IP telephones at a far flung website join a conference, their RTP streams are sent across the WAN to the convention bridge. The convention bridge, no matter if the usage of utility or hardware materials, mixes the bought audio streams and then sends again three exciting unicast audio streams to the IP telephones over the IP WAN. The conference bridge removes the receiver's voice from his or her pleasing RTP flow so that the user doesn't journey echo because of the delay of traversing the WAN link and mixing RTP audio streams in the conference bridge.
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Centralized convention resources cause bandwidth, delay, and capability challenges in the voice network. every G.711 RTP circulation requires eighty kbps (plus the Layer 2 overhead), leading to 240 kbps of IP WAN bandwidth consumption by using this voice conference. If the convention bridge had been now not discovered on the different side of the IP WAN, this site visitors would no longer need to traverse the WAN hyperlink, leading to much less extend and bandwidth consumption. If the far flung web site had a CUCM place configuration that resulted in calls with the G.729 codec returned to the main web page, the software conferencing substances of CUCM would now not be 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. native hardware conference elements would eliminate this want. All centrally found media resources (song On hang [MOH], annunciator, convention bridges, videoconferencing, and media termination aspects) endure similar bandwidth, prolong, and aid exhaustion challenges.
When deploying CUCM in multisite environments, centralized CUCM-primarily based capabilities are accessed over the IP WAN. Affected capabilities consist of the following:
figure 1-three shows a Unified Communications network in which the main web page is related to a faraway web page through a centralized call-processing ambiance. The leading web page is additionally connected to a far off cluster via an intercluster trunk (ICT) representing a distributed call processing atmosphere. The mixture of both centralized and disbursed call processing represents a hybrid call-processing model wherein small sites use the CUCM resources of the main web page, but large far off places of work have their own CUCM cluster. On the backside left of determine 1-three is a SIP trunk, typically over a Metro Ethernet connection to a web Telephony provider provider (ITSP). The improvement of the SIP trunk is that the ITSP provides the gateways to the PSTN as an alternative of your providing gateways at the main web site.
click for a bigger photograph
An IP WAN outage in determine 1-three will cause an outage of name-processing capabilities for the far flung web page related in a centralized style. The remote cluster will no longer suffer a name-processing outage, but the remote cluster will not be in a position to dial the main web site over the IP WAN right through the outage. Mission-crucial voice functions (voice mail, interactive voice response [IVR], and so forth) observed on the leading web page may be unavailable to any of the other sites during the WAN outage.
If the ITSP is the use of the same links that enable IP WAN connectivity, all calls to and from the general public switched cell community (PSTN) will also be unavailable.
A deployment like the one proven in figure 1-3 is regarded badly designed on account of the lack of IP WAN and PSTN backup.
Dial plan challenges
In a multisite deployment, with a single or dissimilar CUCM clusters, dial plan design requires the honour of a number of considerations that do not exist in single-site deployments:
each interior directory number relates to a hard and fast-length PSTN quantity: in this case, each inner directory number has its personal dedicated PSTN number. The listing number can, however doesn't ought to, in shape the least-big digits of the PSTN quantity. In countries with a fixed numbering plan, such because the North American Numbering Plan (NANP), this continually skill that the four-digit workplace codes are used as inner listing numbers. If these are not entertaining, digits of workplace codes or administratively assigned website codes should be would becould very well be brought, leading to 5 or more digits being used for interior directory numbers.
one other answer is to not reuse any digits of the PSTN number but to without difficulty map each internally used listing number to any PSTN quantity assigned to the business. in this case, the inside and external numbers should not have anything in average. If the internally used directory quantity fits the least-big digits of its corresponding PSTN number, massive digits can also be set at the gateway or trunk. additionally, established external telephone number masks, transformation masks, or prefixes will also be configured. here's genuine as a result of all interior directory numbers are changed to fully qualified PSTN numbers in the identical approach. yet another instance is if the inner directory number is composed of elements of the PSTN number and administratively assigned digits such as web page codes plus PSTN station codes, or diverse ranges, such as PSTN station codes 4100 to 4180 that map to directory numbers 1100 to 1180, or completely unbiased mappings of interior directory numbers to PSTN numbers. in that case, one or greater translation guidelines should be used for incoming calls, and one or more calling party transformation suggestions, transformation masks, exterior telephone quantity masks, or prefixes must be configured.
type of quantity (TON) in ISDN: The calling quantity (the computerized number Identification [ANI]) of calls being received from the PSTN can be represented in other ways:
- As a seven-digit subscriber number- As a ten-digit quantity, including the area code- In overseas format with the country code in entrance of the enviornment code
To standardize the ANI for all calls, the layout that's used ought to be typical, and the number has to be modified consequently.
Overlapping and nonconsecutive numbers
In figure 1-four, Cisco IP telephones at the leading site use directory numbers 1001 to 1099, 2000 to 2157, and 2365 to 2999. on the faraway web site, 1001 to 1099 and 2158 to 2364 are used. These listing numbers have two issues. First, 1001 to 1099 overlap; these directory numbers exist at each websites, so they aren't entertaining right through the complete deployment. This reasons an issue: If a user within the remote site dialed simplest the four digits 1001, which phone would ring? This problem of overlapping dial plans has to be addressed by means of digit manipulation. in addition, the nonconsecutive use of the range 2000 to 2999 (with some duplicate numbers at the two websites) would require a big variety of extra entries in name-routing tables because the tiers can infrequently be summarized through one (or a couple of) entries.
click for a larger graphic
fastened versus variable-length numbering plans
a set numbering plan facets fastened-length area codes and local numbers. An open numbering plan facets variance in length of area code or local quantity, or both, within the country.
desk 1-1 contrasts the NANP and a variable-length numbering plan -- Germany's numbering plan during this instance.
desk 1-1component Description mounted Numbering Plan (NANP) Variable-size Numbering Plan (Germany) country code A code of 1 to three digits is used to attain the selected mobilephone equipment for every nation or particular carrier. achieve the E.164 usual from http://itu.org to peer all overseas country codes. 1 49 area code Used inside many countries to route calls to a particular metropolis, vicinity, or special carrier. reckoning on the nation or place, it could possibly even be referred to as a numbering plan enviornment, subscriber trunk dialing code, country wide vacation spot code, or routing code. Three digits Three to 5 digits Subscriber number Represents the selected cell quantity to be dialed, however it does not include the nation code, enviornment 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 initial digits to be dialed in a home name, earlier than the enviornment code and the subscriber number. 1 0 access code a number it really is historically dialed first "to get out to the PSTN," used in PBXs and VoIP systems. nine0 overseas prefix The code dialed earlier than a global quantity (nation code, enviornment code if any, and then subscriber number). 011 00 or + (+ is used by mobile phones)
The NANP PSTN quantity is 408-555-1234, DID is not used, and all calls positioned to the main site are dealt with by means of an attendant. there's a far off website in Germany with the E.164 PSTN quantity +forty nine 404 13267. 4-digit extensions are used at the German area, and DID is allowed as a result of digits may also be added to the PSTN number. When calling the German office attendant (not realizing a specific extension), U.S. clients would dial 9-011-forty nine-404-13267. word how the + is changed by using the overseas prefix 011 and the entry code 9. If the phone with extension 1001 should be known as directly, 9-011-forty nine-404-13267-1001 must be dialed.
Variable-size numbering, E.164 addressing and DID
determine 1-5 illustrates an illustration in which the main site with CUCM resides in the U.S. and a far flung web page without CUCM resides in Germany. The NANP PSTN quantity in the U.S. is 408-555-1234. note that DID is not used, as a result of all calls placed to the leading web page are handled by an attendant. A far flung web site in Germany has PSTN quantity +49 404 13267. four-digit extensions are used at the German location, and DID is allowed as a result of digits can also be introduced to the PSTN number. When calling the German workplace attendant (no longer realizing a particular extension), U.S. clients would dial 9-011-49-404-13267. If the telephone with extension 1001 should still be referred to as without delay, 9-011-49-404-13267-1001 has to be dialed.
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The good judgment of routing calls by means of CUCM over the WAN or during the PSTN is correctly transparent to the phone user.
Optimized name routing and PSTN backup
There are two methods to store charges for PSTN calls in a multisite deployment:
within the instance shown in figure 1-6, a call from Chicago to San Jose could be routed as follows:
click on for a bigger picture
If the WAN had been unavailable for any reason before the name, the Chicago Gateway would should be effectively configured to route the call with the acceptable digit manipulation during the PSTN at a potentially higher toll can charge to the San Jose PSTN phone.
NAT and safety considerations
In single-web page deployments, CUCM servers and IP telephones continually use inner most IP addresses as a result of there isn't any need to talk with the outside IP world. NAT is not configured for the cell subnets, and assaults from the outside are unattainable. In multisite deployments, although, IP security (IPsec) virtual inner most community (VPN) tunnels can be used between sites. The VPN tunnels enable most effective intersite communication; entry to the blanketed inside networks isn't feasible from the backyard—handiest from the other site during the tunnel. for this reason, attacks from the outdoor are blocked on the gateway. To configure IPsec VPNs, the VPN tunnel should be configured to terminate on the two gateways within the distinct sites. every now and then here's not feasible; for example, both websites could be under distinctive administration, or perhaps security guidelines do not allow the configuration of IPsec VPNs.
In this sort of case, or when connecting to a public carrier equivalent to an ITSP, NAT needs to be configured for CUCM servers and IP telephones. Cisco calls this Hosted NAT Traversal for Session Border Controllers.
In determine 1-7, company A and business B both use IP network 10.0.0.0/eight internally. To communicate over the internet, the inner most addresses are translated into public IP addresses. enterprise A uses public IP community A, and business B makes use of public IP network B. All CUCM servers and IP phones can be reached from the cyber web and communicate with each and every different.
As quickly as CUCM servers and IP phones can be reached with public IP addresses, they're discipline to attacks from the outside world, introducing knowledge security issues.
click on for a larger pictureThis get first appeared at searchunifiedcommunications.com
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