avvid update
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1Presentation_ID 1999, Cisco Systems, Inc.
Cisco AVVIDCisco AVVIDArchitecture for Voice, Video, and Integrated Data
Emerging Voice Technologies
Architecture for Voice, Video, and Integrated Data
Emerging Voice Technologies
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2Presentation_ID 1999, Cisco Systems, Inc.
Owen [email protected] [email protected]
Consulting Engineer
Voice & MultiService Technologies
Cisco Enterprise Line Of Business Technical Consultancy
Consulting Engineer
Voice & MultiService Technologies
Cisco Enterprise Line Of Business Technical Consultancy
2Presentation_ID 1999, Cisco Systems, Inc.
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www.cisco.comZermatt 2000 - Slide number 3 of 65
AgendaAgenda
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
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DVVI ConsolidationDVVI Consolidation
PCs
Distribution
ServerFarm
WiringCloset
Voice Mission-Critical
Application
VideoSurveillance
Mission-Critical
Application
Web Servers
Four Different Traffic Types - all Mission Critical
PCs, Voice, Video, Business AppsPCs, Voice, Video, Business Apps
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Key Technology Requirements for IP TelephonyKey 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 spaceis fundamental
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Cisco AVVIDp An End-to-End ArchitectureCisco AVVIDp An End-to-End Architecture
Softphone
MessageMessage
ServerServer
TelephonyTelephony
ApplicationApplication
ServersServersPagingPaging
ServerServerDirectoryDirectory
ServersServers
ContentContentServerServer
ContentContentServersServers
IP PhonesPCs
Video
Distributed
Adaptive
Distributed
Adaptive
Gateway Switch Router
Cisco IP Fabric
Intelligent Network Services
Platforms
InfrastructureClients Applications
Open
Manageable
Open
Manageable
MessageMessage
ServersServers
CallManagerCallManager
ServersServers
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PSTN
What do I need for IP Telephony?What do I need for IP Telephony?
Phones
Supporting, H.323, SIP or a low weight stimulusresponse 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
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Where do all these bits fit?Where do all these bits fit?
WAN
PBX PBX
PSTNCisco IP Phones:12SP+/30VIP
Next Generation
Application
Integration:Voice Mail
Dialing Plans
SupplementaryServices
Cisco
CallManager
LAN Switches provide
QoS, inline Power and
Switched Infrastructure
GatewaysProvide Analog or Digital Access
and QoS support
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Open, Standards-BasedOpen, Standards-Based
Gateway
Cisco
Directory
Server
CallManager
Legacy
PBX
Voice-Enabled
Router
Cisco IP Phones
QoS Enabled
Catalyst Switch
Application
Servers
IP
PSTN
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Open System ApplicationsNewWorld Ecosystem
Open System ApplicationsNewWorld Ecosystem
IP
GatewayGatewayCallManagerCallManager QoS
EnabledCatalyst
Switch
QoS
EnabledCatalyst
Switch
Application
Servers
Application
Servers
Telekol
Picazo
Meet Me ConferencePSTN
Voice Mail/UMS from
Amteva Acquisition
Call Center Applications
from GeoTel Acquisition
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12Presentation_ID 1999, Cisco Systems, Inc.
The ComponentsFor Call Manager Release 2.4
The ComponentsFor Call Manager Release 2.4
12F0_4339_c1 1999, Cisco Systems, Inc.
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.cisco.comZermatt 2000 - Slide number 13 of 65
Cisco AVVID IP PhonesCisco 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
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.cisco.comZermatt 2000 - Slide number 14 of 65
and if you have a lot of legacy phonesand 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
Stillwaiting for BU decision on product
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.cisco.comZermatt 2000 - Slide number 15 of 65
Cisco AVVID PSTN GatewaysCisco 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.
PBXPBX
P
S
T
N
PBXPBX
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.cisco.comZermatt 2000 - Slide number 16 of 65
53005300 72027202
72047204
72067206
75057505
75077507
75137513
17501750
Cisco Multiservice GatewaysCisco Multiservice Gateways
Performance
2600260036203620
36403640
36603660
38103810
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.cisco.comZermatt 2000 - Slide number 17 of 65
Cisco AVVID Call ManagerCisco 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
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.cisco.comZermatt 2000 - Slide number 18 of 65
Cisco AVVID WEB Attendant ConsoleCisco AVVID WEB Attendant Console
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Cisco AVVID CDR LinkCisco 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
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20Presentation_ID 1999, Cisco Systems, Inc.
QoS For IP TelephonyWithout it nothing works !
QoS For IP TelephonyWithout it nothing works !
20F0_4339_c1 1999, Cisco Systems, Inc.
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.cisco.com
Zermatt 2000 - Slide number 22 of 65
.1
G.711G.711
G.729(a)G.729(a)
G.723.1G.723.1
64k + Header64k + Header
8k + Header8k + Header
6.3k /5.3kk + Header6.3k /5.3kk + Header
79.5k (on Ethernet)79.5k (on Ethernet)
20k/10k (on PPP)20k/10k (on PPP)
18k/8k (on PPP)18k/8k (on PPP)
So HowMuch BandwidthDoes A Voice Call Really Need ?
So HowMuch BandwidthDoes A Voice Call Really Need ?
Voice calls without Compressed Real Time Protocol can take up
as much as 80Kon a WAN linkwhen using G.711
Administrator can defines which CODEC in regions
Currently no low-bitrate conferencing
*Current Phones Use These Codecs
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.cisco.com
Zermatt 2000 - Slide number 23 of 65
ChallengeChallenge Cisco SolutionsCisco Solutions
Packet Residency
Slow Link Freeze-out byLarge Packets
Interleaving
FRF.12, MLPPP, IP MTU Size Reduction,Faster Link
Bandwidth Consumption
Header Size on Low
Bandwidth Links
Compression
Codecs, RTP Header Compression, Voice
Activity Detection
WAN
Oversubscription, Bursting
Traffic Management
Router Traffic Shaping to CIR, High PriorityPVC, Data Discard Eligibility
Congestion
Delay and Delay Jitter
Intelligent Queuing
WFQ, IP Precedence, RSVP,Priority Queuing
VoIP Low Speed Link (
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24Presentation_ID 1999, Cisco Systems, Inc.
Some RecommendationsSome Recommendations
24
F0_4339_c1 1999, Cisco Systems, Inc.
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.cisco.com
Zermatt 2000 - Slide number 25 of 65
Network Infrastructure - LANNetwork 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
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.cisco.com
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Network Infrastructure - WANNetwork 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)
WAN
Router
Multilayer
Campus
Multilayer
Campus
Router
CallManagerCallManager
Campus
Not as CriticalInitially
Must Be Switched
WAN Edge
A Must
QoS Startsin the WAN
WAN Backbone
A Must
Often Overlooked
or Misunderstood
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.cisco.com
Zermatt 2000 - Slide number 27 of 65
Policy NetworkingPolicy Networking
Telecommuters
MobileUsers
BranchOffices
Enterprise Policy
QoSSecurity
Campus
Partners
Define policies for applications
and users
Distribute policy bindings
QoS Policy Servers, Security
Network enforcement nodes
Enable integrated control over
network resources
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28Presentation_ID 1999, Cisco Systems, Inc.
IP Phone RoadmapRemember IP Devices That Support Voice
IP Phone RoadmapRemember IP Devices That Support Voice
28F0_4339_c1 1999, Cisco Systems, Inc.
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.cisco.com
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Not Committed
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.cisco.com
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Cisco 7960 IP Phone- Needs Call Manager Release 3.0
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 andApplications
Built-in Headset Connection
10/100 BaseT - 3 Port Switch
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.cisco.com
Zermatt 2000 - Slide number 31 of 65
Cisco AVVID IP SoftphoneCisco AVVID IP Softphone
True mobility
Work from home but answer as if youare 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
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32Presentation_ID 1999, Cisco Systems, Inc.
Call Manager Release 3.0Planned FCS Q2 CY 2000
Call Manager Release 3.0Planned FCS Q2 CY 2000
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.cisco.com
Zermatt 2000 - Slide number 33 of 65
Very Strict Guidelines For 3.0Very 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
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.cisco.com
Zermatt 2000 - Slide number 34 of 65
Call Manager Release 3.0Call 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 EnhancementsToll Restriction User Classes
Distributed Region Number Management
Regionalized Gateway & Tail End Hop Off
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Call Manager Release 3.1Call 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
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36Presentation_ID 1999, Cisco Systems, Inc.
Distributed Call ProcessingDistributed Call Processing
36F0_4339_c1 1999, Cisco Systems, Inc.
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San Jose
Cluster
SanJoseA
SanJoseB
SanJoseCSanJoseD
SanJoseE
CM 3.0 ClustersCM 3.0 Clusters
(N + 1) Scheme that provides
Enhanced Redundancy
Scalability
CM 3.0Up to 5 CM(s) Per Cluster
2500 Devices Per CM
512M Memory Required
10,000 IP Phones Per Cluster
(5-1) * 2500 = 10,000
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How Do Cluster Communicate InternallyHow Do Cluster Communicate Internally
Database used is SQL 7.0 (+ SQL 7.0 SP1)
1 Publisher Per Cluster
Remaining CMs are Subscribers
(N -1) TCP connections 25 CMs = 24 Connections
All Configuration changes made on Publisher
Call Manager (Real Time Data)
Fully Meshed.
(N * (N - 1)) TCP connections
5 CMs = (5 x 4) = 20 Connections
25 CMs = (25 x 24) = 600 Connections
Real Time Data - Phone / Gateway Registrations etc
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How Do Clusters CommunicateInternally
How Do Clusters CommunicateInternally
Sequel
Database
Intra Cluster
Signaling = Full Mesh
Publisher
Subscriber
Subscriber Subscriber
Subscriber
7830
7820
78207820
7820
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N+1 exampleN+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
SanJoseD SanJoseC
SanJoseB
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N+1 exampleN+1 example
SanJoseD Powered off. The devices lose their connection to CM SanJoseD.
SanJoseA
SanJoseE
SanJoseD SanJoseC
SanJoseB
X
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SanJoseA
SanJoseE
SanJoseD SanJoseC
SanJoseB
X
N+1 exampleN+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.
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44Presentation_ID 1999, Cisco Systems, Inc.
Catalyst EnhancementsCatalyst Enhancements
44F0_4339_c1 1999, Cisco Systems, Inc.
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Catalyst IP Telephony EnhancementsCatalyst 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
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47Presentation_ID 1999, Cisco Systems, Inc.
So How Does This All WorkSo How Does This All Work
47F0_4339_c1 1999, Cisco Systems, Inc.
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Cisco AVVID : Basic UnderstandingCisco AVVID : Basic Understanding
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PBXPBX
P
ST
N
PBXPBX
IP IP
IPIP
70325 70330
70330 goes off-hook
sends OH message to CM
CM tells to give dial tone
70330 dials 70325
CM tells 70325 you have
a call with extension 70330
CM tells 70330 IP address
of 70325
70330 and 70325 have a
conversation on how best
to make the call work
70330 and 70325 have
conversation
CM drops out of picture --
no longer involved
when the conversation
is over 70330 and 70325
notify CM they are on-hook
Cisco AVVID : IP to IP CallCisco AVVID : IP to IP Call
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PBXPBX
P
ST
N
IP IP
IPIP
70330
70330 goes off-hook
sends OH message to CM
CM tells to give dial tone
70330 dials number in
PSTN space
Cisco AVVID : IP to PSTN CallCisco AVVID : IP to PSTN Call
Based on Dial Plan, CM
determineswhich Gateway
and which port is to act
CM notifies gateway
CM notifies 70330 the IP
address of Gateway
70330 contacts Gateway
Call parameters set up
call is completed through
PSTN
CM releases involvement
with call
when the conversation
ends, 70330 and Gateway
notify CallManager that
connection is over
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CiscoCallManager
Cisco uOneMessaging Server
IMAP4
Message
Store
52015202
CFNA-->5050
1-Call
Setup
2-E.164 lookup
3-Call
Setup4-Alerting
(Ring)
4-Alerting
(Ringback)
5-No Answer
8-Media
Connect
50506-Call Setup
7-Auto-Offhook
11-Greeting/tone
playback
(SMTP)
13-Message Waiting
14-Message
Waiting
11-Greeting/tone
playback (SMTP)
12-Record
Message (RTP) 12-Record
Message
(SMTP)
LDAP v3
Directory
9-Get
Subscriber
10-Set
Subscriber
All signaling and media over IPAll signaling and media over IP
Cisco CallManager Voice Mail callcontrol 5201 dials 5202 - fwd to VMCisco CallManager Voice Mail call
control 5201 dials 5202 - fwd to VM
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52Presentation_ID 1999, Cisco Systems, Inc.
Application IntegrationApplication Integration
52F0_4339_c1 1999, Cisco Systems, Inc.
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Application IntegrationApplication 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
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uOneCheck My
Calendar
Cell / PDA
IP Network
If Its a Customer, Ring Me Wherever!
If Sister Calls and I Am in a Meeting, Send to Unified Messaging
If Its the Boss, and Im not in a Meeting, Ring Me but Dont Call Cell
Virtual Intelligent AssistantVirtual Intelligent Assistant
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55Presentation_ID 1999, Cisco Systems, Inc.
2.4 Design Guide2.4 Design Guide
55F0_4339_c1 1999, Cisco Systems, Inc.
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Design CaveatsDesign 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 use80kbps for G.711)
Call Manager registers EACH GKControlled H.323 device as a
separate VoIP-GW with Gatekeeper
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Router/GW
PSTN
V
Site ASite B
Site C
IP WAN
Router
IP WAN
Router
Call
Manager
V
V
Call
ManagerCall
Manager
Call Manager at each site
200 IP Phones
100 Voice Mail Boxes
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
=
Amteva Gate Server
Message Store
Directory
Isolated DeploymentsIsolated Deployments
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Router/GW
IP WAN(Primary Voice Path)
PSTN(Secondary
Voice Path)V
Site A
Site B
Site C
IP WAN
Router
IP WAN
Router
Call
Manager
V
V
Call
Manager
IOS Gatekeeper forAdmission Control
Call Manager at Site
200 IP Phones
100 Voice Mail Boxes
IOS Gateways Required
G.711/80kbps for ALLALL IP WAN voice calls
Maximum 10 IP WAN sites
Transparent PSTN Fallback if WAN Unavailable
No Voice Mail Networking
Multi-site WANDistributed Call Processing
Multi-site WANDistributed Call Processing
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Router/GW
IP WAN
Centralized
Call Manager
PSTNV
Location
Hub
Location 2
Location 3
V
IP WAN
Router
V
IP WANRouter
Multi-site WANCentralized Call Processing
Centralized Dial Plan
200 IP Phones
100 Voice Mail Boxes
Admission Control - Location BW Limits
Remotes mustmust use Selsius GWs due to MTP
128kbps Minimum (Plan for 80kbps per Call evenwith Selsius Gateways)
No Service if WAN down (Unless Dial Backup)
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Call Manager
2.4 (Required)
Voice/Voice/
EmailEmail
Messaging
Server
uOne GateServer
(4.1E)
Message Store/Directory
Directory
uOne 4.1E1. 100 Voice mail Boxes or less2. 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
All components on
MCS-7830
Skinny Station Protocol
Gateserver registers
as an IP Phone
MCS-7830
Voice MessagingAll in one Entry Package
Voice MessagingAll in one Entry Package
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uOne Entry Edition IncludesuOne 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
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63Presentation_ID 1999, Cisco Systems, Inc.
3.0 Design Guide3.0 Design Guide
63F0_4339_c1 1999, Cisco Systems, Inc.
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PSTN
V
Site ASite B
Site C
V
V
A
A
A
CM 3.0 Isolated DeploymentsCM 3.0 Isolated Deployments
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)
G.711 - Makes no sense to compress calls
GW Selection based on PSTN signaling
MTP Not Required
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PSTN
V
Site A
Site B
Site C
V
V
A
A
A
IP WAN(Primary Voice Path)
Gatekeeper
CM 3.0 Multi-Site WanDistributed Call processing
CM 3.0 Multi-Site WanDistributed Call processing
Call Manager Cluster per site
10,000 IP Phones Per Cluster
Max 10 Clusters (5 Redundant) - IOS GK for CAC
G.711 or G.729 for WAN (C6K DSP Farm required for G.729 & Conference)
GW Selection based on PSTN signaling
MTP Not Required
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Location Based Admission Controland Cluster Interaction
Location Based Admission Controland Cluster Interaction
Not Supported SupportedSupported
Cluster1CMA
Cluster1CME
Cluster1CMD Cluster1CMC
Cluster1CMB
Location 1 Location 2
One CCM in a clusterOR up to
Three CCMs 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
BA
Location 1 Location 2
Primary
CM
Backup
CM
Locations
WAN Cluster
Remote
Sites
Central
Site
Remote
Sites
Central
Site C
Backup
CM
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Router/GW Router/GW
Call
Manager
IP WAN
Compressed Call Leg in the IP WAN
DSP Resource at Central Site
uOne
Gateserver
DSP = DSP FarmG.711 Call Leg
Compressed Call Leg
Supported/Committed for 3.0
DSP
3.0 Solution SetNo DSP Farm or Cisco CallManager at remote sites
3.0 Solution SetNo DSP Farm or Cisco CallManager at remote sites
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GW
Router/GW
IP WAN
Centralized
Call Manager
VPSTNV
Site1 Site2
Dial 9 for Local PSTN GW
Dial 9 for Local PSTN GW
Partition
1. Devices with similar reachability
characteristics
2. Items placed in Partition:
IP Phones, DNs, 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
3.0 Dial Plan Enhancements -Partitions
Centralized Call Manager but Distributed Dial Plan
3.0 Dial Plan Enhancements -Partitions
Centralized Call Manager but Distributed Dial Plan
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36003600
IP WAN
337-1XXX 447-2XXX
VV
Dial Plan - Robust Digit ManipulationDial Plan - Robust Digit Manipulation
PSTNVoice
OverflowUser dials
447-2222
1111 2222
2222 Presented
to Call Manager
36003600
IP WAN
337-1XXX 447-2XXX
VVPSTN
VoiceOverflowUser dials
447-2222
1111 2222
2222 Presented
to Call Manager
Call ManagerCall Manager
strips 447 for IP WANstrips 447 for IP WAN
Call ManagerCall Managerprepre--pends 1610 for PSTNpends 1610 for PSTN
Call Placed Across IP WANCall Placed Across IP WAN
Call rejected Across IP WAN, takes PSTNCall rejected Across IP WAN, takes PSTN
Gatekeeper
Gatekeeper
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IOS GW
Call Manager
with
Amteva Gateserver
1. Allows for IOS GW/Call Manager Integration without MTP (Greater Scalability)
2.2. MGCP support for Analog IOS GW interfaces ONLY (FXS & FXO)MGCP support for Analog IOS GW interfaces ONLY (FXS & FXO)
3. Fax will require G.711 Operation with MGCP3. Fax will require G.711 Operation with MGCP
PSTNV
Design Characteristics
IOS GW
Call Manager
with
Amteva Gateserver
PSTNV
Initial Call - Direct from GW to IP Phone
No MTP Required
Call Transfer - Allow Supplementary Services
SkinnyStation
MGCP VoicePath
MGCPElimination of MTP
MGCPElimination of MTP
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IOS GW
Cisco CallManager
with
Amteva Gateserver
1. Allows for IOS GW/Call Manager Integration without MTP (Greater Scalability)2. IOS gatewayswith Digital Interfaces without MTP3. Minimum IOS 12.0(7)T
PSTNV
Design Characteristics
IOS GW
Cisco CallManager
with
Amteva Gateserver
PSTNV
Initial Call - Direct from GW to IP Phone
No MTP Required
Call Transfer - Allow Supplementary Services
SkinnyStation
H323v2 VoicePath
Elimination of MTPH.323v2with Open Closed Logical Channel
Elimination of MTPH.323v2with Open Closed Logical Channel
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IOS Gatekeeper
V VRouter/GW Router/GW
CallManager
Cluster
CallManager
Cluster
IP WAN
Design Characteristics
Compressed WAN Call Leg across IP WAN
Gatekeeper used for Admission Controlwith PSTN Fallback
Cisco uOne
Gateserver
Cisco uOne
Gateserver
V = DSP FarmG.711 Call Leg
Compressed Call Leg
Site A Site B
Call Manager and DSP Farm at each siteCall Manager and DSP Farm at each site
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Summary & ConclusionSummary & Conclusion
LAN & WAN Infrastructure needs QoS
Without it give up and go home !
Network Design should support redundancy
Voice systems do today !
Ciscos has solutions to support Voice/Video/Data enabled
networks
Cisco AVVID Packet Voice Is A Real Solution !
Cisco AVVID ArchitecturePowerful, flexible IP architecture that supports Intelligent Integrated Applications
Cisco AVVID Advanced Applications
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75P t ti ID
Thank-you.Thank-you.
Owen Bridle
Cisco Voice & MultiService Technologies Consulting Engineer
Enterprise LOB
Owen Bridle
Cisco Voice & MultiService Technologies Consulting Engineer
Enterprise LOB