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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 quality 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 suggest specific options.
picking out concerns 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. fastened versus variable-length numbering plans8. Variable-size numbering, E.164 addressing and DID9. Optimized name 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 fault tolerance. each packet is processed one after the other in an IP community, every so often inflicting distinctive packets in a communications circulation to take distinct paths to the destination. The distinctive paths in the community may also have a special amount of packet loss, delay, and lengthen version (jitter) as a result of bandwidth, distance, and congestion differences. The vacation spot must be capable of get hold of packets out of order and resequence these packets. This challenge is solved by the use 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 ideal load-balancing paradigm, however ensures that each and every IP flow (voice call) takes the same course.
Bandwidth is shared by varied clients and purposes, whereas the amount of bandwidth required for a person IP movement varies enormously all through brief lapses of time. Most statistics purposes are very bursty, whereas Cisco true-time audio communications with RTP use the same continual-bandwidth movement. The bandwidth purchasable for any application, together with CUCM and voice-bearer traffic, is unpredictable. right through top intervals, packets need to be buffered in queues waiting to be processed on account of network congestion. Queuing is a time period that any person who has ever skilled air flight is widely wide-spread with. in case you arrive at the airport, you must get 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 standard on these sorts of hyperlinks, and you need to deal with them by imposing QoS. with out the use of QoS, voice packets adventure extend, jitter, and packet loss, impacting voice high-quality. it is critical to correctly configure Cisco QoS mechanisms conclusion to end right through the community for relevant audio and video performance.This chapter became excerpted with permissions from imposing Cisco Unified Communications manager, part 2, by means of Chris Olsen, Copyright 2008. For extra assistance about this publication and other titles, please discuss with Cisco Press.
all the way through peak intervals, packets can't be despatched automatically on account of interface congestion. as a substitute, the packets are quickly stored in a queue, waiting to be processed. The amount of time the packet waits within the queue, called the queuing delay, can differ tremendously in response to community situations and traffic arrival rates. If the queue is full, newly received packets cannot 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 incessant (sampling is each 20 milliseconds via default), but records purposes are bursty and grasping. Voice for this reason is subject 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 way of a metropolitan-enviornment network (MAN) comparable to Metro Ethernet. Bandwidth on WAN hyperlinks is restricted and comparatively expensive. The goal is to make use of the obtainable bandwidth as correctly as possible. useless traffic should still be removed from the IP WAN hyperlinks via content material filtering, firewalls, and access manage lists (ACL). IP WAN acceleration methods for bandwidth optimization should still be regarded as well. Any duration of congestion might outcome in carrier degradation until QoS is deployed all the way 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 use 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 be converted to bits to figure 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 way 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 have been enabled). In evaluation to records software packets, voice packets are very small (60 bytes for G.729 and 200 bytes for G.711 with the default 20-ms sampling price).
In figure 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 be a part 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 interesting RTP circulation so that the consumer does not experience echo 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 found 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 place configuration that resulted in calls with the G.729 codec again to the main website, the software conferencing resources of CUCM would 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. local hardware convention supplies would get rid of this need. All 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 small websites use the CUCM supplies of the leading web page, but large far off workplaces have 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 figure 1-three will cause 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 all over the outage. Mission-critical voice applications (voice mail, interactive voice response [IVR], and the like) determined at the main web page could be unavailable to any of the other sites all the way through the WAN outage.
If the ITSP is the use of the same hyperlinks that enable IP WAN connectivity, all calls to and from the public switched telephone community (PSTN) will also be unavailable.
A deployment like the one shown in determine 1-three is regarded badly designed as a result of the shortcoming of IP WAN and PSTN backup.
Dial plan challenges
In a multisite deployment, with a single or distinctive CUCM clusters, dial plan design requires the distinction of a number of concerns that do 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 Plan (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 be delivered, resulting in five or extra digits getting used for inside listing numbers.
one other answer 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 do not need anything else in typical. If the internally used listing number matches the least-significant digits of its corresponding PSTN quantity, huge digits may also be set at the gateway or trunk. additionally, commonplace exterior cell quantity masks, transformation masks, or prefixes can also be configured. here is proper as a result of all 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 have to be used for incoming calls, and one or greater calling birthday party transformation rules, transformation masks, external mobile quantity masks, or prefixes ought to be configured.
type of quantity (TON) in ISDN: The calling quantity (the computerized number Identification [ANI]) of calls being bought from the PSTN will also be represented in alternative ways:
- As a seven-digit subscriber number- As a ten-digit quantity, 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 that's used have to be favourite, and the quantity must be transformed hence.
Overlapping and nonconsecutive numbers
In determine 1-four, Cisco IP phones at the leading website use 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 have 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 be addressed by digit manipulation. moreover, the nonconsecutive use of the range 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 be summarized by means of one (or a few) entries.
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fixed versus variable-length numbering plans
a fixed numbering plan elements fastened-size area codes and native numbers. An open numbering plan facets variance in length of area code or native quantity, or both, within the country.
desk 1-1 contrasts the NANP and a variable-size numbering plan -- Germany's numbering plan during this illustration.
desk 1-1component Description fixed Numbering Plan (NANP) Variable-length Numbering Plan (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 all 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 also be called a numbering plan 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 be dialed, nevertheless it doesn't consist of the nation code, enviornment code (if applicable), foreign prefix, or trunk prefix. Three-digit exchange code plus a four-digit station code Three or greater digits Trunk prefix The initial digits to be 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 get out to the PSTN," utilized in PBXs and VoIP methods. 90 international prefix The code dialed before an international quantity (nation code, area 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 all 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 be 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 still be referred to as without delay, 9-011-49-404-13267-1001 must be 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 all 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 also be 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 be referred to as without delay, 9-011-49-404-13267-1001 has to be dialed.
click for a larger graphic
The good judgment of routing calls by means of CUCM over the WAN or in the course of the PSTN is accurately clear 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 name from Chicago to San Jose would be routed as follows:
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If the WAN were unavailable for any intent before the name, the Chicago Gateway would need to be effectively configured to route the call with the appropriate 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 use 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 also be 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 be configured to terminate on both gateways within the diverse websites. on occasion this is now not viable; for example, both websites may be 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 be configured for CUCM servers and IP phones. Cisco calls this Hosted NAT Traversal for Session Border Controllers.
In figure 1-7, enterprise A and company B each use IP community 10.0.0.0/8 internally. To speak 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. All CUCM servers and IP telephones may also be reached from the information superhighway and talk with each and every different.
As soon as CUCM servers and IP phones may also be 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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