Deploying Voice over IP in Campus Environments
Download
Report
Transcript Deploying Voice over IP in Campus Environments
Cisco AVVID
Architecture for Voice, Video, and Integrated Data
‘Emerging Voice Technologies’
Presentation_ID
© 1999, Cisco Systems, Inc.
1
Owen Bridle
[email protected]
Consulting Engineer
Voice & MultiService Technologies
Cisco Enterprise Line Of Business Technical Consultancy
Presentation_ID
© 1999, Cisco Systems, Inc.
2
Agenda
• Consolidation
• Cisco IP Telephony
– IP Telephony Components
• But First …. Quality Of Service (QoS)
• IP Phone Roadmap
• Call Manager 3.0
– Distributed Call Processing
– Catalyst Enhancements
• So how does this all work then
• Cisco AVVID Application Integration
• Design Guide
• Summary
Zermatt 2000 - Slide number 3 of 65
www.cisco.com
DVVI Consolidation
Distribution
Wiring
Closet
Voice
Server
Farm
MissionCritical
Application
PCs
Zermatt 2000 - Slide number 4 of 65
Video
Surveillance
MissionCritical
Application
Web Servers
Four Different Traffic Types - all Mission Critical
PCs, Voice, Video, Business Apps
www.cisco.com
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
Zermatt 2000 - Slide number 5 of 65
www.cisco.com
Cisco AVVID IP Telephony
Presentation_ID
F0_4339_c1
© 1999, Cisco Systems, Inc.
6
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
Zermatt 2000 - Slide number 7 of 65
Router
• Open
• Manageable
www.cisco.com
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
PSTN
– PBX and PSTN connectivity
• Applications and Call Processing
– CallManager, voice mail, IVR, etc.
• Network infrastructure
– Routers, switches, wire, WAN services
Zermatt 2000 - Slide number 8 of 65
www.cisco.com
Where do all these bits fit?
PSTN
Cisco IP Phones:
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
Zermatt 2000 - Slide number 9 of 65
www.cisco.com
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
Zermatt 2000 - Slide number 10 of 65
Voice-Enabled
Router
www.cisco.com
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
QoS
Enabled
Catalyst
Switch
Telekol
Picazo
Zermatt 2000 - Slide number 11 of 65
www.cisco.com
Gateway
IP
The Components
For Call Manager Release 2.4
Presentation_ID
F0_4339_c1
© 1999, Cisco Systems, Inc.
12
Cisco AVVID 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
Zermatt 2000 - Slide number 13 of 65
www.cisco.com
…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
Still waiting for BU decision on product
– Ericsson MD110, Business Phone Dialog 3000
Zermatt 2000 - Slide number 14 of 65
www.cisco.com
Cisco AVVID PSTN Gateways
• All Cisco IOS/CatOS Platforms
– 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
Zermatt 2000 - Slide number 15 of 65
www.cisco.com
P
S
T
N
Cisco Multiservice Gateways
7513
7206
7204
3660
3640
2600
1750
Zermatt 2000 - Slide number 16 of 65
5300
7507
7505
7202
3620
3810
www.cisco.com
Performance
Cisco AVVID 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
Zermatt 2000 - Slide number 17 of 65
www.cisco.com
Cisco AVVID WEB Attendant Console
Zermatt 2000 - Slide number 18 of 65
www.cisco.com
Cisco AVVID CDR Link
• Solution would not be complete without billing records.
– CDR saved to hard disk in CSV format
– File can be exported to traditional systems
Zermatt 2000 - Slide number 19 of 65
www.cisco.com
QoS For IP Telephony
Without it nothing works !
Presentation_ID
F0_4339_c1
© 1999, Cisco Systems, Inc.
20
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
Zermatt 2000 - Slide number 21 of 65
www.cisco.com
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 “regions”
– Currently no low-bitrate conferencing
G.711
64k + Header
79.5k (on Ethernet)
.1
G.729(a)
8k + Header
20k/10k (on PPP)
G.723.1
6.3k/5.3kk + Header
18k/8k (on PPP)
*Current Phones Use These Codecs
Zermatt 2000 - Slide number 22 of 65
www.cisco.com
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
Traffic Management
WAN
Oversubscription, Bursting
Zermatt 2000 - Slide number 23 of 65
Router Traffic Shaping to CIR, High Priority
PVC, Data Discard Eligibility
www.cisco.com
Some Recommendations
Presentation_ID
F0_4339_c1
© 1999, Cisco Systems, Inc.
24
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
Zermatt 2000 - Slide number 25 of 65
www.cisco.com
Network Infrastructure - WAN
• Redundancy via Routing Protocols
– Need to reroute quickly, 3 minutes is too long for voice
• Sufficient WAN bandwidth
– Whilst data may ride for free it still needs bandwidth
• Quality of Service
– 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
Zermatt 2000 - Slide number 26 of 65
Multilayer
Campus
WAN Edge
“A Must”
QoS Starts
in the WAN
www.cisco.com
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
Zermatt 2000 - Slide number 27 of 65
www.cisco.com
IP Phone Roadmap
Remember IP Devices That Support Voice
Presentation_ID
F0_4339_c1
© 1999, Cisco Systems, Inc.
28
Not Committed
Zermatt 2000 - Slide number 29 of 65
www.cisco.com
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
Zermatt 2000 - Slide number 30 of 65
www.cisco.com
Cisco AVVID 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
• TAPI Application
• Can control associated IP
Phone
• Directory Integration
• Call Manager Release 3.0(2)
– Currently works with 2.4
Zermatt 2000 - Slide number 31 of 65
www.cisco.com
Call Manager Release 3.0
Planned FCS Q2 CY 2000
Presentation_ID
F0_4339_c1
© 1999, Cisco Systems, Inc.
32
Very Strict Guidelines For 3.0
• No Beta allowed in the field until 3rd week in March
– EFT starts end of February beginning of March 2000
• Must be co-ordinated by the EMEA Tiger Team
– Contact Adrian Brookes ([email protected]) or Eric Chapat ([email protected])
• No SE training until week beginning 27th March 2000
– First course at the Technical Forum in Brussels
– Limited availability, open to Tiger Team members only
• Why ?
– Major changes in 3.0
– Just completed, but many changes requested
– FCS looks like May 2000
Zermatt 2000 - Slide number 33 of 65
www.cisco.com
Call Manager Release 3.0
• 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
Zermatt 2000 - Slide number 34 of 65
www.cisco.com
Call Manager Release 3.1
• 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
– Personal Call Flow agent enhancements
• E911 device auto-location and notification
– E911 call control upon disconnect
• Digit Analysis Translation Tables
• Malicious Call Trace CDR Support
Zermatt 2000 - Slide number 35 of 65
www.cisco.com
Distributed Call Processing
Presentation_ID
F0_4339_c1
© 1999, Cisco Systems, Inc.
36
CM 3.0 Clusters
• (N + 1) Scheme that provides
– Enhanced Redundancy
– Scalability
• CM 3.0
SanJoseA
– Up to 5 CM(s) Per Cluster
SanJoseE
– 2500 Devices Per CM
San Jose
Cluster
SanJoseB
– 512M Memory Required
• 10,000 IP Phones Per Cluster
– (5-1) * 2500 = 10,000
Zermatt 2000 - Slide number 37 of 65
www.cisco.com
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
Zermatt 2000 - Slide number 38 of 65
www.cisco.com
How Do Clusters Communicate
Internally
Sequel
Database
Intra Cluster
Signaling = Full Mesh
7830
Publisher
7820
7820
Subscriber
7820
Subscriber
7820
Subscriber
Zermatt 2000 - Slide number 39 of 65
Subscriber
www.cisco.com
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
Zermatt 2000 - Slide number 40 of 65
www.cisco.com
SanJoseC
N+1 example
• SanJoseD Powered off. The devices lose their connection to CM SanJoseD.
SanJoseA
SanJoseE
SanJoseB
X
SanJoseD
Zermatt 2000 - Slide number 41 of 65
www.cisco.com
SanJoseC
N+1 example
• The devices re-home to other
call managers, which then
replicate new route and
registration information to
each other.
SanJoseA
SanJoseE
• 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.
Zermatt 2000 - Slide number 42 of 65
www.cisco.com
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)
Zermatt 2000 - Slide number 43 of 65
www.cisco.com
Primary
Secondary
Last Resort
Catalyst Enhancements
Presentation_ID
F0_4339_c1
© 1999, Cisco Systems, Inc.
44
Next Generation IP Phone
Line Powered Line Cards - Cisco 7960
• 10/100 switch port
• Catalyst Ethernet Line Card
• Web Client
• Catalyst 3500, 4K, 6K
– Uses Pins 1,2,3,6
• IP Prec = 5, DSCP = EF, Cos = 5
• 802.1Q VLAN Support
• Cat5k Patch Panel
– Uses Pins 4,5,7,8
• Can reclassify 802.1p
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
Zermatt 2000 - Slide number 45 of 65
www.cisco.com
Catalyst IP Telephony Enhancements
• Catalyst 6XXX switch family blades
– 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
– IOS Based Gateway only based on 2600 router platform
Zermatt 2000 - Slide number 46 of 65
www.cisco.com
So How Does This All Work
Presentation_ID
F0_4339_c1
© 1999, Cisco Systems, Inc.
47
Cisco AVVID : Basic Understanding
Zermatt 2000 - Slide number 48 of 65
www.cisco.com
Cisco AVVID : 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
• sends
a
call
OHextension
message
70330”
to CM
is
overwith
70330
and 70325
conversation on how best
•CM
•notify
CM
CMdrops
tells
tells
70330
out
give
of
IP
dial
picture
address
tone-CM to
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
Zermatt 2000 - Slide number 49 of 65
70330
IP
www.cisco.com
PBX
Cisco AVVID : IP to PSTN Call
• when
70330
CM
call
Based
notifies
is the
goes
completed
on conversation
Dial
off-hook
70330
Plan,
through
the
CM
IP
• address
determines
sends
PSTN70330
OH
ofmessage
Gateway
which
Gateway
to CM
ends,
and Gateway
• notify
and
70330
CM
tells
releases
which
contacts
to port
give
involvement
is
Gateway
dial
tothat
tone
act
CallManager
• connection
70330
Call
CM
withnotifies
parameters
call
dials is
number
gateway
set up
in
over
PSTN space
IP
IP
70330
IP
Zermatt 2000 - Slide number 50 of 65
IP
www.cisco.com
PBX
P
S
T
N
Cisco CallManager Voice Mail call
control 5201 dials 5202 - fwd to VM
Cisco
CallManager
Cisco uOne
Messaging Server
All signaling and media over IP
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
Zermatt 2000 - Slide number 51 of 65
5202
CFNA-->5050
www.cisco.com
LDAP v3
Directory
Application Integration
Presentation_ID
F0_4339_c1
© 1999, Cisco Systems, Inc.
52
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
Zermatt 2000 - Slide number 53 of 65
www.cisco.com
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
Zermatt 2000 - Slide number 54 of 65
www.cisco.com
2.4 Design Guide
Presentation_ID
F0_4339_c1
© 1999, Cisco Systems, Inc.
55
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
Zermatt 2000 - Slide number 56 of 65
www.cisco.com
Isolated Deployments
Site A
Call
Manager
=
Site B
Amteva Gate Server
Message Store
Directory
Call
Manager
Router/GW
V
V
IP WAN
Router
PSTN
Call Manager at each site
Call
Manager
V
200 IP Phones
100 Voice Mail Boxes
IP WAN
Router
GW Selection based on PSTN signaling
Zermatt 2000 - Slide number 57 of 65
–
PRI - DT24+/DE-30+ or AS5300
–
T1/E1 CAS - 2600/3600/7200
–
MTP Required for IOS Gateways - Min 12.0(7)T
www.cisco.com
Site C
Multi-site WAN
“Distributed Call Processing”
Site A
Router/GW
Call
Manager
PSTN
(Secondary
Voice Path)
V
V
IP WAN
Router
Site B
IP WAN
(Primary Voice Path)
Call
Manager
Call Manager at Site
IOS Gatekeeper for
Admission Control
200 IP Phones
100 Voice Mail Boxes
V
IP WAN
Router
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
Zermatt 2000 - Slide number 58 of 65
www.cisco.com
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
IP WAN
Router
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)
Zermatt 2000 - Slide number 59 of 65
www.cisco.com
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
Zermatt 2000 - Slide number 60 of 65
www.cisco.com
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”
Zermatt 2000 - Slide number 61 of 65
www.cisco.com
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
Zermatt 2000 - Slide number 62 of 65
www.cisco.com
3.0 Design Guide
Presentation_ID
F0_4339_c1
© 1999, Cisco Systems, Inc.
63
CM 3.0 Isolated Deployments
A
A
V
V
Site A
PSTN
Site B
• Call Manager Cluster at each site
A
– 10,000 IP Phones Per Cluster
V
– 2500 per Cal Manager
– No Limit to Number of Clusters (Each Cluster is isolated PBX)
• G.711 - Makes no sense to compress calls
• GW Selection based on PSTN signaling
Site C
– MTP Not Required
Zermatt 2000 - Slide number 64 of 65
www.cisco.com
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)
A
Gatekeeper
– 10,000 IP Phones Per Cluster
– Max 10 Clusters (5 Redundant) - IOS GK for CAC
V
• G.711 or G.729 for WAN
– (C6K DSP Farm required for G.729 & Conference)
• GW Selection based on PSTN signaling
Site C
– MTP Not Required
Zermatt 2000 - Slide number 65 of 65
www.cisco.com
Location Based Admission Control
and Cluster Interaction
One CCM in a cluster…OR up to
Three CCM’s in a Cluster Providing:
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
Cluster1CMA
Central
Site
Cluster1CME
Cluster1CMB
Cluster1CMD
Central
Site
Primary
CM
A
Backup
CM
Backup
CM
B
C
Cluster1CMC
Remote
Sites
“Locations
WAN Cluster”
Remote
Sites
Location 1
Location 2
Location 1
Not Supported
Zermatt 2000 - Slide number 66 of 65
Location 2
Supported
www.cisco.com
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
Zermatt 2000 - Slide number 67 of 65
www.cisco.com
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
Calling Search Space
Partition
1. Devices with similar “reachability
characteristics”
2. Items placed in Partition:
IP Phones, DN’s, Gateways + Route
Patterns
Zermatt
2000 - Slide number 68 of 65
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
www.cisco.com
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
Zermatt 2000 - Slide number 69 of 65
PSTN
“Voice
Overflow”
www.cisco.com
V
3600
2222 Presented
to Call Manager
MGCP
Elimination of MTP
Skinny
Station
MGCP
Call Manager
with
Amteva Gateserver
Voice
Path
Call Manager
with
Amteva Gateserver
V
PSTN
IOS GW
V
PSTN
IOS GW
Initial Call - Direct from GW to IP Phone
Call Transfer - Allow Supplementary Services
No MTP Required
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
Zermatt 2000 - Slide number 70 of 65
www.cisco.com
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
IOS GW
Initial Call - Direct from GW to IP Phone
No MTP Required
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
Zermatt 2000 - Slide number 71 of 65
PSTN
www.cisco.com
Call Manager and DSP Farm at each site
Compressed Call Leg
G.711 Call Leg
V
= DSP Farm
Site B
Site A
IOS Gatekeeper
CallManager Cisco uOne
Gateserver
Cluster
Cisco uOne CallManager
Cluster
Gateserver
IP WAN
Router/GW
Router/GW
V
V
Design Characteristics
Compressed WAN Call Leg across IP WAN
Gatekeeper used for Admission Control with PSTN Fallback
Zermatt 2000 - Slide number 72 of 65
www.cisco.com
Summary & Conclusions
Presentation_ID
F0_4339_c1
© 1999, Cisco Systems, Inc.
73
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
Zermatt 2000 - Slide number 74 of 65
www.cisco.com
Thank-you.
Owen Bridle
Cisco Voice & MultiService Technologies Consulting Engineer
Enterprise LOB
[email protected]
Presentation_ID
© 1999, Cisco Systems, Inc.
75