CIPT2 Notes

Table of Contents

Quality issues

during congestion packet drops occur
voice large amount of small packets (vs data small amount of large packets)
consider putting DSPs local to site (DSP farm)

conference resources are initiated based on who initiates call

Availability

Paging, services, other centralized concernes

Dial Plan Issues

Overlapping DNs
Nonconsecutive DNs
Variable len numbering and intterdigit timeout handling
optimized call routing

toll bypass
tail-end hop-off
PSTN backup

Various PSTN reqs in various countries

Access scodes for PSTN, national , and international dialing
Number Presentation (ISDN TON)
Consider reformatting caller ID

Scalability

IME (intercompany media engine) encourages use of IP for calls rather then PSTN

overlap sending and receiving

Issues caused by different PSTN dialing

diff ways to store of config PSTN desintations

speed dials
fast dials
addr book entries
call lists
AAR targets
Call Forward destinations

stored numbers can be used at multiple sites (countries) because of roaming users using local PSTNGWs

Extnesion Mobility
Cisco Device Mobility
PSTN backup
TEHO and LCR (Least cost routing)

Scalability

static cfg for multi sites is very complex
centralized H.323 GK o SIP network services offer dial plan
Call Control Discovery (CCD) allows dynamic learning of dial plans

NAT and Security Issues in multisite environments

Muttisite Deployment Solutions

QoS CAC RTP Header Compression, Local Media Resources
SRST, pSTN Backup, MGCP Fallback
Access and site codes, digit transofmraiton
CUBE, SBC

Options to reduce BW

Lo bw codecs on IP WAN
RTP header compression
local annunciators or disabling remote annunciators
local conf bridges
local MTPs,
local transcoders, mixed conf bridges
local MOH or multicast MOH
CAC

Codec used depends on region cfg on CUCM

Cisco phones can force use of equal or lower bw codec
e.g. g711 specified, but g722 negotiated between phones(?!?)

Transcoder CFG

hw does transcoding

multicast MOH from Branch Router flash

MOH capabilities SRST
IP phone cfged toto use CUCJM multicast MOH
CUCM MOH server cfged for mx-hops 1
CUCM MOH and branch router use same multicast addr and port number
CUCM signals MOH svr addr and port number to phone
CUCM MOH svr packets dropped at WAN because max-hops vall TTL exceeded
SRST rtr gens multicast MOH stream with same multicast addr
IP phones listes to signaled addr and port and plays received stream

Alternate to max-hops ACL on local router blocking access across

CAC

Availability options

CFUR
AAR - kicks in when CAC or SIP pre-option says not enough BW
device mobility

Globalized Call Routing

In transit, number tranlsatedto e.164
locally phone number localized
Globalize coming into CM, Localize going out...

Dial plan components in multisite deployment

Dial Plan Component	IOS GW	CUCM
endpoint addr	ephone-dn, dynamic POTS, dial pears	DN
digit manipulation	voice translation profiles using prefix, digit-strip, forward-digits, and num-exp commands	Translation patterns, route patterns, route lists, significant digits, ccalled- and calling-party transformations, incomning called- and callin-party settings
calling privileges	COR and COR lists	Partitions, CSSes, time schedules, time periods, FACs
Call coverage	Dialpeers, call apps ephone hunt groups	line groups hunt lists, hun t

Security

Cube

media flow through vs media flow around
RSVP and SIP preconditions
Adaptive Security Agent (?) ASA integration

Multisite Connection

h323
MGCP
SCCP

Trunks

H323
SIP
QSIG over H323 or SIP works but only basic call flow

MGCP re-establish (new / bkup CM) causes reset of trunk
Cisco prefers you run a ICT or H.225 (now a days H.225) through GK
Now-a-days use SIP trunks (including for Extension Mobility)

H.323 Trunk Comparison

	Nongatekeeper Controlled ICT	Gatekeeper COntrolled ICT	H225 Trunk

MGCP Notes

turn off auto-updating of mgcp gws

leave off ccm-manager config server <ipaddr> and ccm-manager config

H323
Need both commands:
h323-gateway voip bind srcaddr <ipaddr>
h323-gateway voip interface

GK-Controlled ICT and H225 Trunk Cfg Overview

Trunk Types Used by Special apps

Extension Mobility Cross Clusters

Only SIP

Cisco Call Control

SIP Trunk Config

Always set up SIP Profiles & SIP Security Profiles per SIP Trunks - safer

CUCM nongatekeep ICT Config

Dial Plans

Implementing access and site codes

allows routing independient of DNs
solves overlapping and nonconsecutive DNs

Implementing PSTN access

simple prioritized list of GWs for all PSTN access
TEHO

Iimplementing PSTN backup

Route lists and groups for path selection

1st choice: on-net
2nd choice: off-net

Uniform (short) length dial plan (e.g. 4 digits) doesn't scale very well
Variable length dial plan (access code + site code + xxxx)
Fixed dial plan (use e.164 or nanp number with translation patterns)

+0... e.164 numbers are significant for (Cisco) 'Locally Significant' or (ITU) 'reserved code 0' phone numbers
http://cisco.com/go/srnd
features and services guide
administration guide

Numberless enterprise (key on phone connects you to Unity Connection which speech recognizes the name you're trying to call). Or Jabber, or IP Communication)

URI and/or number can show up on line on phone

SAF with CCD

advertise longest number you can

Access and Site Codes

Modify Calling Number on way out of local phone switch

USE E.164 OR NANP FOR COMPHEHENSIVE DIAL PLAN

Selective PSTN Breakout Cfg Options

GW selection by CSS

each site uses device specific CSS
Site specific rt pattern i smatched
route pattern refers to site specific route list
route list includes site-specific rout group
route group refers to site-specific GWs

GW selection by local route group feature

matched route pattern reers to sys wide route list
route list refers to Standard Local Route Group
Device pool of claling device is cfged with route group to be used
eliminnates the need for multi (site-specific) route lists, route patterns, partitions, CSSes
preferrred option since intro of local route grps (CUCM 7)

TEHO issues

PSTN number of originating site at TEHO GW
Replacing PSTN number of originating site by PSTN number of TEHO
Use Local Route Groups (N*N or N+N effort to configure routes)

Globalized Call Routing Characteristics

Use e.164 as base number
localize going out gateways and/or coming in gateways
localize for convenience of user
Use transformations (like translations...)
called number / called party transformation css for gw and dp
TEHO route - second route is local gateway

Localized call egress at phones

Use transformations with e164 for teho, aar, etc.
WHERE DO TRANSFORMATIONS GET APPLIED

Centralized Call Processing Redundancy Implementation

MGCP Fallback
Cisco Unified SRST
Cisco Unified SIP SRST
CUCM Express in SRST mode

SRST Router/Reference

Phone knows about SRST based on Device Pool
If no SRST gw configured, will try to use default router

Can stop this by using loopback for srst gw

SRST Timing

PUT IN TIMERS

MGCP Fallback

MGCP GWs reg with CUCM
MGCP gws exchange keepalive msgs with central CUCM GW across WAN
CUCM is MGCP call agent
If WAN link fails, MGCP GWs lose contact with CUCM
Tries to connectg to fallback
falls back to default application (H.323 (default) or SIP)
switchback - don't set to graceful (waits for all calls to complete).

alternates are after active calls have finished, after fixed amount of time, at a fiexed time of day

e.164 is supported but have to issue special command SRST 8.0 or greater

COMMAND

Dial Plan requirements

keep an eye on 9+ vs non 9+ for regular vs SRST mode...
Use CFUR to route over PSTN when WAN is down

to reduce impact of routing loops, service parameter limits the number of CFUR hops percall
CFUR for Ext Mobility lines should always point to voicemail to avoid routing loops
Use Globalized Call routing with Standard Local Route Groups

COR

Create tags
Create List of tags
Compare incoming dialpeer COR list ot outgoing dial-peeer COR list. If they match, restrict it.

Setting up SRST

Create new SRST ref on CUCM

CUCM Admin/System/SRST

Assign SRST ref to dev pool

CUCM Admin / System / Device Pool
Assign 'SRST Reference (Disable, Use Default GW, or 1 of the SRST cfgs)

setup gw

call-manager-fallback
ip source-address ip-address [port port] [any-match | strict-match]
max-dn max-directory-numbers [dual-line] [preference preference-order]
max-ephones max-phones
limit-dn {phone type} max-lines
keepalive seconds
ccm-manager fallback-mgcp
call application alternate Default
service alternate Default
Max hops
CUCM Admin / System / Service Parameter / Cisco Call Manager / Clusterwide Parameters (Feaature - Forward)
Max Forward Hops to DN

dialplan-patern tag pattern extension-length length [extension-mattern extension-pattern] [no-reg] [demote]

int s0/0/0:23
	isdn negotiate-bchan (Enables the router to accept a B channel that is different from
	the B channel requested in the outgoing call-setup message and
	specifies the cause codes for which the call is reattempted.)
application
	global
	service alternate Default (specifies that default voice application takes over if
	MGCP call agent is unavailable. Allows fallback to H.323
	or SIP. Local dial peers will be considered for call routing).
ccm-manager fall-back-mgcp (enable srst for MGCP conrtrolled ports)

voice translation-rule 2
	rule 1 /^
	(4...\)/ /466\1/
voice translation-profile Internal
	translate called 2
	
no telephony-service
call-manager-fallback (enter CM fallback cfg mode)
	ip source-addr &lg;ipaddr> (gw source addr)
	max-ephones <num>
	max-dn 12 dual-line
	limit-dn <phonetype> <numDNs>
	system message primary Help Me - Network is down
	secondary-dialtone 9 (play dialtone after dialing 9)
	translation-profile incoming internal

dial-peer voice 85101 voip
	destination-pattern 851....
	port 0/0/0:23
	forward-digits all
	prefix 401

Timing

CUCM SRST GW/RTR IP Phone
 <---TCP Keepalive def 30 seconds-->
	WAN Fails
	After 3 missed keepalive msgs
	try secondary CM (~ 60 seconds?)
	After 3 missed keepalive msgs
	try secondary CM (~ 60 seconds?)
	<--SRST Reg--
	10-20 sec
	SRST pulls ip phone cfg
	<--TCP Keepalive def 30 seconds--
	WAN Restored
	120 second switchback timer
 <---Phone re-registers-------------

CME in SRST Mode

emulates a key system / pbx
runs on IOS
administered by gui or cli
call transfer, paging, intercom, call coverage
call park, moh, multicast moh
hunt groups, basic acd
adhoc conf
night bell night svc cafw
ext mobility presence

telephony-service
    max-ephones
    max-dn
    ip source-address
    create cnf-files

ephone-dn 7 (this is line 7)
    number 3002
ephone 3
    mac-addr 1234.5678.2345
    type 7960
    button 1:6 (button 1 uses line 7)

tftp-server flash:SCCP45.9-0-2SR1S.loads

telephony-service
	load 7945 SCCP45.9-0-2SR1S

voice moh-group 1
	moh flash:moh1.au
	description MOH: customer services
	multicast moh 239.1.1.1 port 16384
	extension-range 1000 to 1099
	extension range 2000 to 2099
telephony-service
	moh-file-buffer 5000
	moh flash:default.wav
	multicast moh 239.1.1.3 port 16384

SRST Notes

can hybrid manual/automatic configuration of ephones
if automatic learned can set up ephone-dn templates

CHART FOR Phone Provisioning Options
SNAP SN Auto Provisioning features

Phone reg proc

rt/gw searches for existing ephone with MAC addr

if found, ephone must be configured with DN or auto assignment must be enabled
if not found, rtr searches for existing ephone-dn that matches IP phone DN (learned by SNAP)

if found, ephone added that refers to matched ephone-dn
if not found, ephone added that refers to newly created ephone-dn autoconfigured by SNAP

whever ephone added by SRST, srst ephone template applied
whever ephone-dn added by SRST, srst ephone-dn template applied

srst mode auto-provision {all | dn | none}
 (only learns phone first time)
srst dn line-mode [dual | single}
srst dn template template-tag
srst ephone template  template-tag
srst ephone description string

Enhanced SRST needs Unified Message G (UMG)

Managing Bandwidth

Low bw codecs on WAN usage
RTP header compression (as part of QoS link eficiency mechnaisms
local media resources such as conference bridges
transcoders
multicast moh from branch router flash

Region settings

region settings applied to device pool
when setting system up from scratch

set up phones, gateways, trunks, media resources in different regions

bandwidth default between - g729, within - g711

Local Conf Bridge

person initiating conference dictates conf resources used based on their phones config
Media Resource Manager tries to balance calls out on all the resources it knows about

Transcoder implementation

lo bw codecs to be used on wan, but 1 or both endpoints do not support lo bw codecs

affected endpoint uses hi bw codec towards transcoder
transcoder chges voice stream from hi bw to lo bw codec
lo bw codec sent to other device or transcoder over IP WAN

add transcoder resource in CUCM
cfg transcoder resource in Cisco IOS Software
Cfg mMRGs
Cfg MRGLs
Assign MRGLs to devices

CFG IOS
voice-card 0 
	dspfarm
	dsp services dspfarm

sccp local fastethernet0/0
sccp ccm 10.1.1.1 identifier 1 version 7.0+
sccp

sccp ccm group 1
	associate ccm1 priority 1
	associate profile 1 register HW1_XCoder (name to register ith CUCM)

dspffarm profile 1 transcode
	codec g711ulaw
	codec g711alaw
	max sessions 1
	assciate application SCCP
	no shutdown

MOH Implementation from BR flash implementation

bw consider

acl block particular multicast addr across WAN
OR block all multicast
limit TTL/hops

works only with multicast MOH
based on mOHcapabiliteis of SRST
CUCM must be conffiugred to use multicast MOH
IP phone is not aware that it listens to locally generated MOH
stream generated by MOH svr is prevented from reaching IP WAN
Identical stream is generated locally at branch sites
br touger/gw can stream up to six MOH file

only g711 supported
each stream can be selectively enabled for multicast

up to six different MOH sources supoprted per br router/gw

Copy MOH Cfg on IOS routers

enable multicast routing on rtr

ip multicast-routing

!blocking multicast 
ip access-list extended drop-moh
	deny ip any host 239.1.1.1 range 16384 18385
	permit ip any any

interface FastEthernet 0/0
	description HQ
	ip address 10.1.1.111 255.255.255.224
	ip pim sparse-dense-mode

interface Serial0/1
	description WAN
	ip addr 10.1.3.233 255.255.255.224
	ip pim sparse-dense-mode
	ip access-group drop-moh out

call-manager-fallback
	max-ephones 1
	max-dn 2
	ip source address <ipaddr>
	moh moh-file.au
	multicast moh 239.1.1.1 port 16384

Cfg MOH Audio sources for Multicast MOH on CM

Media Resources / Music on Hold Audio Source

Check 'Allow Multicasting'

Media Resources / Fixed MOH Audio Source

Check 'Allow Multicasting''

Media Reosources / Music on Hold Server

Check Enable Multicast Audio Sources on MOHH Server
cfg multicast ip addr and port and increment mulitasst on port number or ip address
set the number of hops for the Audio sources (files).

Media Resources / Media Resource Group

add MOH resource to MRG and MRGL

CAC

Locations based
RSV Enabled Locations
Autommated Alternate Routing
SIP Preconditions
H.323 Gatekeeper CAC

Location based

calls between locations that exceed calculated bw assigned get blocked

system / location

name loc
assign audio bw

Assign location to device pool and/or phone

RSVP enabled location chars

based on CUCM locations
allows RSVP to be enabled bewteen pairs of locations
Need RSVP agents

using MTPs on routers

Topology aware

works with with all topology (full mesh, partial mesh, hub and spoke
adapts to net chg

link failurs, backup links, load share paths

based on IOS standard RSVP
IP network between RSVP agents is RSVP enabled
each interface is conifgured with max bw to be reserved by RSVP
RSVP is not enabled on any hop in the path, the apporpriate link is ignored by CAC algorithm
IntServ and Diffserv models used

RSVP only for CAC control plane
LLQ for QoS data plane

Call set up only after successful RSVP CAC

Configure RSVP svc params

default interlocation RSVP policy (n reservation, optional, mandatory (both audio and video), mandatory (video desired vid call can proceed as an audioonly call if reservation for autdio streram succeeds but a reservation for the vid streaa does not succeed)
RSVP Retry timer
mandatory RSVP Mid-Call Error Handle Option deifne wheter call retry becomes best oeffort or fails
Mandatory RSVP Mid-Call Retry Counter

cfg RSVP agents in IOS

dspfarm profile 1 mtp
    codec pass-through
    rsvp
    maximum sessions software 20
    associate application SCCP
interface serial0/1
    ...
    ip rsvp bandwidth 40

Add RSVP agents to CUCM
enable RSVP beteween loc pairsCfg MRGs and MRGLs
assign MRGLs to devices

AAR chars

reroutes calls denied by CAC
Use AAR css to elevate permissions that might not be otherwise possible
AAR has its own digit prefix setting if you want to use it...

AAR does not support CTI route points or extension mobility

process

cfg aar service parameters

put msg on screen via service params

cfg css and partitions
cfg aar groups
cfg phones for AAR (config slide on exam)

SIP Preconditions

based on rfc 3312 - integration of resource mgmt and SIP
CUCM currently supports RSVP only
ADD NOTES

H323 CAC

Device Mobility

Devices settings that get modified by Device Mobility settings.

Region - Codec
Location - BW
SRST ref
AAR group
CSS
MRGs and MRGLs
Other settings

2 types of phones can be aplied by Dev MObility

roaming sensitive settings

local route group
date/time group
region
srst reference
mMRGL
location
network local
physical locations
device mobility group

device mobility-related settings:

Device Mobility CSS
AAR CSS
AAR Group
Calling Party Transformation CSS

Can leave roaming settings blank, and will not change
Calling privelages get preserved but local vs. ld changes with dev mobility

line css never modified
device icss is modifid when device roams between physical locations (same DMGs)

when roaming settings over ride phone settings

Cfg elements used by Dev Mobility

Device Pool
Device Mobilyt info
phsyicalcation
device mobility group

CHARTS SHOWING HOW DEVICE MOBILITY WORKS

Dev Mob Cfg Steps

Cfg phys locs
Cfg dev mob group
Cfg dev pools
Cfg dev mobility infos (IP subnets)
Set Dev Mobility mode by using

CM svc param to set default for all phones
phone cfg window for indifvidual cfg per phone

SAF/CCD

CUCM 8.0+
Can advertise and learn callrouting info

including alternate (PSTN) number
SAF client (call agent) advertises and learns.
SAF forwarder exchang info with other forwarders and downstream to clients
SAF Forwaridng Protocol between forwarders
SAF Client protocol (Forwarder to/from clients)

Works with

CUCM (client)
CUCM Express (client)
SRST
CUBE
IOS GW

Uses features and functions of EIGRP including

bw percent
hellow interval
hold time
split horizon
authenticated updates
incremental updates

run separate eigrp routing process (SAF on that)
2 options for neighbor rleationships

l2 adjacent (multicast and unicast)
non l2 adjacent

functions

client

register to net
pub svcs
sub to svcs
keepalives

forwarder

propagate updates
hellows to other forwarders
propagate updates

processing

admin can block received routes
load balancing occurs for learned routes

rouhnd robin between protocols, among local trunks, and learned remote ip addrs

partitions and CSS

all learned patterns are put into one conigurable partition
all devices should have access to learned routs as neede acceess
aar css to use for pstn backup calls

routes stop - persist route then secondary

implementation

cfg saf forwarders on IOS - specify same AS on all forwarders
cfg trunk profile with ip to be used for call setup
MORE MORE MORE

router eigrp SAF
service-family ipv4 autonomous-system 1
	sf-interface FastEthernet0/0
	topology base
		exit-sf-topology
		external-client HQ_SAF
	exit-service-family
!service-family external-client listen ipv4 5050
	external-client HQ_SAF
		username DAUSERNAME
		password DAPASSWORD

Setup steps on external saf client/cm/CM Express

Cfg SAF security profile
Cfg SAF forwarder
Cfg SAF (SIP or H.323) trunk
Cfg hosted DN group
Cfg hosted DN pattern
Cfg CCD advertising service
Cfg CCD requesting service and partition
Cfg CCD blocked learned patterns (optional)
Cfg CCD feature parameters

USE RTMT TO SEE SAF LEARNED ROUTES