Transcript CME Network Design

Converged Network Design
Agenda
•
•
Consolidation
Cisco IP Telephony
– IP Telephony Components
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•
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But First …. Quality Of Service (QoS)
IP Phone Roadmap
Call Manager 3.0
– Distributed Call Processing
– Catalyst Enhancements
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So how does this all work then
Cisco AVVID Application Integration
Design Guide
Summary
Consolidation
Distribution
Wiring
Closet
Voice
PCs
Server
Farm
MissionCritical
Application
Video
Surveillance
MissionCritical
Application
Web Servers
Four Different Traffic Types - all Mission Critical
PCs, Voice, Video, Business Apps
Key Technology Requirements for IP Telephony
• Reliability
– Dial tone is always there
• Quality of Service
– WAN and LAN, good quality voice always!
• Power & Current Infrastructure Integration
– eg: cabling, integration with current PBX
• Scalability to Large Campus Sizes
– Need to move beyond the current 200 user limit
• Application Integration - Current and Future
– Call processing, Unified Messaging, Call center etc.
integration in the Enterprise space is fundamental
Cisco AVVID  An End-to-End Architecture
Clients
Infrastructure
Applications
Cisco IP Fabric
Intelligent Network Services
CallManager
Servers
Message
Servers
Softphone
Message
Telephony
Server
Application
Servers
Video
Platforms
PCs
IP Phones
Gateway
Switch
• Distributed
• Adaptive
Router
• Open
• Manageable
Content
Content
Server
Servers
Paging
Directory
Server
Servers
What do I need for IP Telephony?
• Phones
– Supporting, H.323, SIP or a low weight stimulus
response protocol such as Skinny e.g. Symbol, Cisco
• Gateways
– PBX and PSTN connectivity
• Applications and Call Processing
– CallManager, voice mail, IVR, etc.
• Network infrastructure
– Routers, switches, wire, WAN services
PSTN
Where do all these bits fit?
Cisco IP Phones:
PSTN
12SP+/30VIP
Next Generation
PBX
Application
Integration:
Voice Mail
Dialing Plans
Supplementary
Services
PBX
Gateways
Provide Analog or Digital Access
and QoS support
WAN
Cisco
CallManager
LAN Switches provide
QoS, inline Power and
Switched Infrastructure
Open, Standards-Based
Application
Servers
Legacy
PBX
Cisco
Directory
Server
PSTN
CallManager
Gateway
QoS Enabled
Catalyst Switch
IP
Cisco IP Phones
Voice-Enabled
Router
Open System Applications—
New World Ecosystem
Call Center Applications
from GeoTel Acquisition
Voice Mail/UMS from
Amteva Acquisition
Application
Servers
PSTN
Meet Me Conference
CallManager
Telekol
Picazo
QoS Enabled
Catalyst
Switch
Gateway
IP
Cisco IP Phones
• Firstly they look like phones…...
• Soft key & display based access
to features and value-added
services
– Programmable soft buttons
– No paper labels, easy set
installation / relocation
• Integral 10BaseT Hub
• Locally powered
• Range of Voice services
– G.711 & G.723.1
…and if you have a lot of legacy
phones
• Support via H.323 & Skinny
• Connect existing circuit
switched phones to an IP
network.
• Supported phones
– Lucent Definity all 8400, 6400 series and Call
Master 3
– Nortel Norstar 7324, 7310, 7208 and 7100,
Meridian M2006, M2008, M2009, M2012, M2216,
and M2616
– Mitel SX-50, SX-200 ,SX-2000 Superset 410, 420,
430 (DNIC), and 4000 series
– Siemens Hicom 110E,150E, 300E All Optiset E,
– Rolm 9751 RP200, RP300, RP400, and RP600
– Ericsson MD110, Business Phone Dialog 3000
Still waiting for BU decision on product
Cisco PSTN Gateways
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All Cisco IOS/CatOS Platforms
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The link from IP to Circuit Switched - H323 Gateway
Both Digital & Analogue support
Support for Supplementary Services (Hold, Transfer, Conference, etc)
Available today…….
PBX
PBX
P
S
T
N
Cisco Multiservice Gateways
7513
7507
7206
7204
3660
3640
2600
1750
5300
7505
7202
3620
3810
Performance
Cisco Call Manager
• Provides intelligent Call Processing & PBX functionality
– Advanced Call Control
– Scalable and on open systems
– Fault tolerant
• Browser Accessed
• Standards based
– H323 and MGCP support
• Application Integration
– Unified Messaging
– Call Centres
Quality of Service - What is it?
• QoS = Preferential treatment
• QoS prioritizes traffic into service levels and
provides preferential treatment to some traffic at
the expense of lower priority traffic
– Needed in both the LAN and the WAN
• All traffic gets the same service
– We now have the possibility of three types of traffic,
all delay sensitive
• A traditional network is best-effort !
– This will not work with a heavily loaded Voice enabled
network
So How Much Bandwidth
Does A Voice Call Really Need ?
• Voice calls without Compressed Real Time
Protocol can take up as much as 80K on a
WAN link when using G.711
• Administrator can defines which CODEC in
G.711
64k + Header
79.5k (on Ethernet)
“regions”
–.1 Currently
low-bitrate conferencing
G.729(a)
8k +no
Header
20k/10k (on PPP)
G.723.1
6.3k/5.3kk + Header
18k/8k (on PPP)
VoIP Low Speed Link (<768 KBPS)
Challenges and Solutions
Challenge
Cisco Solutions
Congestion
Intelligent Queuing
Delay and Delay Jitter
WFQ, IP Precedence, RSVP,
Priority Queuing
Packet Residency
Interleaving
Slow Link Freeze-out by
Large Packets
FRF.12, MLPPP, IP MTU Size Reduction,
Faster Link
Bandwidth Consumption
Compression
Header Size on Low
Bandwidth Links
Codecs, RTP Header Compression, Voice
Activity Detection
WAN
Oversubscription, Bursting
Traffic Management
Router Traffic Shaping to CIR, High Priority
PVC, Data Discard Eligibility
Network Infrastructure - LAN
• Switched Ethernet to the desktop
– No shared flat networks - PLEASE !!
• VLAN support
– Voice in one Data in the other.
• Power capability required
– Phones are powered you know.
• Either in-line or from Power cube
• Redundant Network Design (to address availability)
• Quality of Service
– Bandwidth is not just the answer
– Via Buffer Management and Classification of traffic
Network Infrastructure - WAN
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Redundancy via Routing Protocols
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Sufficient WAN bandwidth
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Need to reroute quickly, 3 minutes is too long for voice
Whilst data may ride for free it still needs bandwidth
Quality of Service
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Without it your are toast
How many of your customers run 100Mb in both the LAN and the WAN ?
Avoiding Loss, Delay, and Delay Variation
(Jitter)
CallManager
CallManager
Router
Router
WAN
Multilayer
Campus
Campus
Not as Critical
“Initially”
Must Be Switched
Multilayer
Campus
WAN Edge
“A Must”
QoS Starts
in the WAN
WAN Backbone
“A Must”
Often Overlooked
or Misunderstood
Policy Networking
• Define policies for
applications and users
• Distribute policy bindings
– QoS Policy Servers, Security
– Network enforcement nodes
Telecommuters
• Enable integrated control over
Campus
network resources
Mobile
Users
QoS
Security
Enterprise Policy
Branch
Offices
Partners
Not Committed
Cisco 7960 IP Phone
- Needs Call Manager Release 3.0
• Professional, Manager
• High or Busy Telephone Traffic
• Six Lines – Mix Directory Numbers or Features
• Full Duplex Handsfree
• Display Area:
– Calling Information, Feature Access Via “Soft
Keys,” Additional Display Area for Value-added
Services and Applications
• Built-in Headset Connection ·
• 10/100 BaseT - 3 Port Switch
Cisco IP Softphone
• True mobility
– Work from home but answer as if
you are in the office.
– Work from the hotel room !
– Now true desktop convergence
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TAPI Application
Can control associated IP Phone
Directory Integration
Call Manager Release 3.0(2)
– Currently works with 2.4
Call Manager Release 3.0
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Distributed Call Processing (N+1)
Enhanced CallManager Database Administration
MGCP - Phase 1
Distributed Call/Single CDR
Online Non-Intrusive Configuration Changes
Telephone UI Software Support
Partitioning Enhancements
– Toll Restriction User Classes
– Distributed Region Number Management
– Regionalized Gateway & Tail End Hop Off
Call Manager Release 3.1
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802.1P/Q & RSVP
Additional Telephone Feature software support
MGCP- Phase 2
Class of Service- User Logon
Class of Service- User Profile download to Telephone upon Login.
Dynamic Comfort Noise adjustment to 7960
OA&M Enhancements for SMB Platforms
Valet/User Model
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E911 device auto-location and notification
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Personal Call Flow agent enhancements
E911 call control upon disconnect
Digit Analysis Translation Tables
Malicious Call Trace CDR Support
CM 3.0 Clusters
• (N + 1) Scheme that provides
– Enhanced Redundancy
– Scalability
• CM 3.0
SanJoseE
– Up to 5 CM(s) Per Cluster
– 2500 Devices Per CM
SanJoseA
San Jose
Cluster
SanJoseB
• 512M Memory Required
• 10,000 IP Phones Per Cluster
– (5-1) * 2500 = 10,000
SanJoseD
SanJoseC
How Do Cluster Communicate
Internally
• Database used is SQL 7.0 (+ SQL 7.0 SP1)
– 1 Publisher Per Cluster
– Remaining CM’s are Subscribers
– (N -1) TCP connections
• 25 CM’s = 24 Connections
– All Configuration changes made on Publisher
• Call Manager (Real Time Data)
– Fully Meshed.
– (N * (N - 1)) TCP connections
• 5 CM’s = (5 x 4) = 20 Connections
• 25 CM’s = (25 x 24) = 600 Connections
– Real Time Data - Phone / Gateway Registrations etc
How Do Clusters Communicate
Internally
Sequel
Database
Intra Cluster
Signaling = Full Mesh
7830
Publisher
7820
7820
Subscriber
7820
Subscriber
7820
Subscriber
Subscriber
N+1 example
• Three devices are homed to
SanJoseD. All nodes in the
network are connected and are
relaying route and registration
information to each other.
SanJoseA
SanJoseE
SanJoseB
SanJoseD
SanJoseC
N+1 example
• SanJoseD Powered off. The devices lose their connection to CM SanJoseD.
SanJoseA
SanJoseE
SanJoseB
X
SanJoseD
SanJoseC
N+1 example
• The devices re-home to other
call managers, which then
replicate new route and
registration information to each
other.
• The devices experience only a
brief outage - (TBD)
– Calls in progress are dropped.
– Gateway calls are dropped.
• Since device operation is
identical, users may not notice
that anything happened.
SanJoseA
SanJoseE
SanJoseB
X
SanJoseD
SanJoseC
Call Manager Groups
• Each device – IP Phone
– Skinny Gateway
• Has a prioritized list of up to 3
Call Managers to which it can
connect.
• This is called a CallManager
Group
• This system is revertive (TBD)
Primary
Secondary
Last Resort
Next Generation IP Phone
Line Powered Line Cards - Cisco 7960
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10/100 switch port
Web Client
IP Prec = 5, DSCP = EF, Cos = 5
802.1Q VLAN Support
Can reclassify 802.1p
• Catalyst Ethernet Line Card
• Catalyst 3500, 4K, 6K
– Uses Pins 1,2,3,6
• Cat5k Patch Panel
– Uses Pins 4,5,7,8
Phone Plugged in
Switch detects IP Phone and applies power
CDP Transaction between Phone and Switch
IP Phone placed in proper VLAN
DHCP request and initialization
Catalyst IP Telephony Enhancements
• Catalyst 6XXX switch family blades
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48 port 10/100 switch module with 48vdc power to phones
• High density DSP resource module (transcoding, mixed codec conferencing)
• 8 port T1 PRI ISDN VoIP gateway module
• 24 port FXS module
• Catalyst 4XXX switch family
–
48 port 10/100 switch module with 48 vdc power
• Modular VoIP gateway with 2 VIC/WIC and one WIC and two fixed FXO/FXS groups (50 pin Telco
connector)
• Catalyst 3500T
• Stackable switch with 48vdc power to phones
• VG200 Gateway
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IOS Based Gateway only based on 2600 router platform
Cisco AVVID : Basic Understanding
IP to IP Call
• 70330
70330
CM
tells
goes
and
70325
70325
off-hook
“you
have
have
when
the
conversation
• 70330 and 70325 have a
conversation
a• call
sends
OHextension
message
70330”
to CM
is
overwith
70330
and 70325
conversation on how best
•CM
• CM
CMdrops
tells
tells
to
70330
out
give
of dial
IP
picture
address
tone-notify
CM they
are
‘on-hook’
to make the call work
no
• 70330
of
70325
longer
dials
involved
70325
PBX
P
S
T
N
IP
IP
70325
IP
IP
70330
PBX
IP to PSTN Call
70330
CM
call
Based
is
notifies
completed
on
goes
Dial
off-hook
70330
Plan,
through
the
CMIP
• when
the
conversation
•address
determines
PSTN
sends70330
OH
of message
Gateway
which
Gateway
to CM
ends,
and Gateway
•notify
and
70330
CM
which
tells
releases
contacts
toport
give
involvement
isGateway
dial
tothat
act
tone
CallManager
•connection
with
70330
Call
CM notifies
parameters
call
dialsisnumber
gateway
set up
in
over
PSTN space
IP
IP
70330
IP
IP
PBX
P
S
T
N
Cisco CallManager Voice Mail call
control
5201
dials
5202
fwd
to
VM
Cisco
Cisco uOne
CallManager
All signaling and media over IP
Messaging Server
13-Message Waiting
2-E.164 lookup
5050
6-Call Setup
7-Auto-Offhook
14-Message
Waiting
3-Call
Setup
1-Call
Setup
4-Alerting
(Ringback)
8-Media
Connect
11-Greeting/tone
playback (SMTP)
10-Set
Subscriber
4-Alerting
(Ring)
12-Record
Message (RTP)
11-Greeting/tone
playback
(SMTP)
9-Get
Subscriber
12-Record
Message
(SMTP)
5-No Answer
IMAP4
Message
Store
5201
5202
CFNA-->5050
LDAP v3
Directory
Application Integration
“When my wife calls to tell me she's going into
labour, it's no big deal for an IP-based switch
to look into my scheduler, find the conference
room I'm in and forward the call there. Getting
a traditional PBX to do that would take a
whole fleet of Andersen consultants.”
Christian Renaud, Cisco EVBU Product Marketing
27th September 1999
Virtual Intelligent Assistant
If Sister Calls and I Am in a Meeting, Send to Unified Messaging
If It’s the Boss, and I’m not in a Meeting, Ring Me but Don’t Call Cell
If It’s a Customer, Ring Me Wherever!
Check My
Calendar
uOne
IP Network
Cell / PDA
Design Caveats
• Maximum 200 users per CM
• No WAN connectivity
• Call Manager cannot use Gatekeeper for Address
Resolution (Does not enhance dial plan scalability)
• No two sites may have same internal dial plan
– Therefore only 1 instance of 1XXX, 2XXX, 3XXX etc
– Limited to 10 Site Deployments
• Call Manager uses 128kbps in the ARQ (Only going to
use 80kbps for G.711)
• Call Manager registers EACH GK Controlled H.323
device as a separate VoIP-GW with Gatekeeper
Isolated Deployments
Site A
Call
Manager
=
Site B
Amteva Gate Server
Message Store
Directory
Call
Manager
Router/GW
V
V
PSTN
IP WAN
Router
Call
Manager
Call Manager at each site
200 IP Phones
100 Voice Mail Boxes
V
IP WAN
Router
GW Selection based on PSTN signaling
–
–
–
PRI - DT24+/DE-30+ or AS5300
T1/E1 CAS - 2600/3600/7200
MTP Required for IOS Gateways - Min 12.0(7)T
Site C
Multi-site WAN
“Distributed Call Processing”
Site A
Router/GW
V
PSTN
(Secondary
Voice Path)
Call
Manager
V
IP WAN
Router
Site B
IP WAN
(Primary Voice Path)
Call
Manager
Call Manager at Site
200 IP Phones
100 Voice Mail Boxes
IOS Gatekeeper for
Admission Control
IOS Gateways Required
G.711/80kbps for ALL IP WAN voice calls
Maximum 10 IP WAN sites
Transparent PSTN Fallback if WAN Unavailable
No Voice Mail Networking
V
IP WAN
Router
Site C
Multi-site WAN
“Centralized Call Processing”
Centralized
Call Manager
Location
Hub
Location 2
V
Router/GW
V
PSTN
IP WAN
Router
IP WAN
Centralized Dial Plan
Location 3
V
200 IP Phones
100 Voice Mail Boxes
Admission Control - Location BW Limits
Remotes must use Selsius GW’s due to MTP
128kbps Minimum (Plan for 80kbps per Call even with Selsius Gateways)
No Service if WAN down (Unless Dial Backup)
IP WAN
Router
Multi-site WAN Topologies
Hub and Spoke
Topologies
Admission Control + Capacity Planning
Ensure voice traffic at every site does not
exceed configured WAN Bandwidth
Minimum requirements for Voice, Video
and Data shall not exceed 75% of link or
VC bandwidth
(Remaining 25% for Routing Protocol updates
and link layer header BW consumption etc.)
Required Link Capacity =
(Min BW for Voice + Min BW for Video + Min BW for Data) / 0.75
Voice Messaging
“All in one” Entry Package
MCS-7830
Message Store/Directory
Messaging
Server
Voice/
Email
Directory
uOne GateServer
(4.1E)
All components on
MCS-7830
Skinny Station Protocol
Gateserver registers
as an IP Phone
Call Manager
2.4 (Required)
uOne 4.1E
1. 100 Voice mail Boxes or less
2. Voice Messaging only
3. Appserver requires Out of Band DTMF and is G.711 only (12.0(7)T)
4. Out of Band DTMF from IOS GW need to be “dtmf-relay h245-alphanumeric”
uOne Entry Edition Includes
• 100 user mailboxes
• 30 minutes storage max per user
• 4 simultaneous streams (ports) of voice
messaging
• OPTIONAL 4 stream (port) upgrade (Maximum
is 8 Ports)
• To go beyond 8 ports or 100 users you will
have to upgrade to a stand alone uOne
Enterprise Version
CM 3.0 Isolated Deployments
A
A
V
V
Site A
PSTN
Site B
• Call Manager Cluster at each site
–
–
–
10,000 IP Phones Per Cluster
2500 per Cal Manager
No Limit to Number of Clusters (Each Cluster is isolated PBX)
A
V
• G.711 - Makes no sense to compress calls
• GW Selection based on PSTN signaling
–
MTP Not Required
Site C
CM 3.0 Multi-Site Wan
Distributed Call processing
A
A
Site A
PSTN
V
V
Site B
IP WAN
• Call Manager Cluster per site
–
–
(Primary Voice Path)
10,000 IP Phones Per Cluster
Max 10 Clusters (5 Redundant) - IOS GK for CAC
A
Gatekeeper
V
• G.711 or G.729 for WAN
–
(C6K DSP Farm required for G.729 & Conference)
• GW Selection based on PSTN signaling
–
MTP Not Required
Site C
Location Based Admission Control
and Cluster Interaction
One CCM in a cluster…OR up to
Three CCM’s in a Cluster Providing:
Central
Site
Cluster1CMA
Cluster1CME
Cluster1CMB
Cluster1CMD
1. All Phones always registered with same Cisco CallManager
2. Achieved by keeping all Phones with same CM Group list order
3. Creation of a “Locations Cluster” of two or three
4. Possible Location status synch during CM failover - self healing
Central
Site
Primary
CM
Backup
CM
A
Backup
CM
B
C
Cluster1CMC
“Locations
WAN Cluster”
Remote
Sites
Remote
Sites
Location 1
Location 2
Not Supported
Location 1
Location 2
Supported
3.0 Solution Set
No DSP Farm or Cisco CallManager at remote sites
Compressed Call Leg
G.711 Call Leg
DSP
= DSP Farm
Supported/Committed for 3.0
uOne
Gateserver
Call
Manager
IP WAN
Router/GW
Router/GW
DSP
Compressed Call Leg in the IP WAN
DSP Resource at Central Site
3.0 Dial Plan Enhancements Partitions
Centralized Call Manager but Distributed Dial Plan
Centralized
Call Manager
Site1
Site2
Router/GW
V
PSTN
V
GW
Dial “9” for Local PSTN GW
IP WAN
Dial “9” for Local PSTN GW
Partition
1. Devices with similar “reachability
characteristics”
2. Items placed in Partition:
IP Phones, DN’s, Gateways + Route
Patterns
Calling Search Space
1. Which Partitions a device may search in for a
dialed number
2. Provides dialing permissions/restrictions
3. Each device “assigned” a Calling Search Space
Dial Plan - Robust Digit Manipulation
337-1XXX
447-2XXX
Gatekeeper
IP WAN
2222
1111
V
3600
User dials
447-2222
Call Manager
strips “447” for IP WAN
V
PSTN
“Voice
Overflow”
3600
2222 Presented
to Call Manager
Call Placed Across IP WAN
Call rejected Across IP WAN, takes PSTN
337-1XXX
447-2XXX
Gatekeeper
IP WAN
2222
1111
V
3600
User dials
447-2222
Call Manager
pre-pends “1610” for PSTN
PSTN
“Voice
Overflow”
V
3600
2222 Presented
to Call Manager
MGCP
Elimination of MTP
Skinny
Station
MGCP
Voice
Path
Call Manager
with
Amteva Gateserver
Call Manager
with
Amteva Gateserver
V
V
PSTN
IOS GW
Initial Call - Direct from GW to IP Phone
No MTP Required
PSTN
IOS GW
Call Transfer - Allow Supplementary Services
Design Characteristics
1. Allows for IOS GW/Call Manager Integration without MTP (Greater Scalability)
2. MGCP support for Analog IOS GW interfaces ONLY (FXS & FXO)
3. Fax will require G.711 Operation with MGCP
Elimination of MTP
H.323v2 with Open Closed Logical Channel
Skinny
Station
H323v2
Voice
Path
Cisco CallManager
with
Amteva Gateserver
Cisco CallManager
with
Amteva Gateserver
V
V
PSTN
IOS GW
Initial Call - Direct from GW to IP Phone
No MTP Required
PSTN
IOS GW
Call Transfer - Allow Supplementary Services
Design Characteristics
1. Allows for IOS GW/Call Manager Integration without MTP (Greater Scalability)
2. IOS gateways with Digital Interfaces without MTP
3. Minimum IOS 12.0(7)T
Call Manager and DSP Farm at each
site
Compressed Call Leg
G.711 Call Leg
V
= DSP Farm
Site B
Site A
IOS Gatekeeper
Cisco uOne
Gateserver
CallManager
Cluster
CallManager
Cluster
IP WAN
V
Router/GW
Router/GW
V
Design Characteristics
Compressed WAN Call Leg across IP WAN
Gatekeeper used for Admission Control with PSTN Fallback
Cisco uOne
Gateserver
Summary & Conclusion
• LAN & WAN Infrastructure needs QoS
– Without it give up and go home !
• Network Design should support redundancy
– Voice systems do today !
• Cisco’s has solutions to support Voice/Video/Data
enabled networks
• Cisco AVVID Packet Voice Is A Real Solution !
• Cisco AVVID Architecture
– Powerful, flexible IP architecture that supports Intelligent
Integrated Applications
• Cisco AVVID Advanced Applications