Cisco AVVID

Architecture for Voice, Video, and Integrated Data

‘Emerging Voice Technologies’

Presentation_ID © 1999, Cisco Systems, Inc. 1

 Owen Bridle
obridle@cisco.com
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

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 DVVI Consolidation

Distribution

Wiring Server
Closet Farm

Voice Mission- Video Mission- Web Servers

Critical Surveillance Critical
Application Application

Four Different Traffic Types - all Mission Critical

PCs
PCs, Voice, Video, Business Apps
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 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

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 Cisco AVVID IP Telephony

Presentation_ID
F0_4339_c1 © 1999, Cisco Systems, Inc. 6
 Cisco AVVID  An End-to-End Architecture

Clients Infrastructure Applications

ICE SUP
R V Cisco IP Fabric POR
SE T
Intelligent Network Services
CallManager
Servers
Softphone Message
Servers

Message
Telephony
Server
Application
Servers
Paging
Directory
Video Server
Servers
Platforms

Content
Content
Server
Gateway Switch Router Servers
PCs IP Phones

• Distributed • Open
• Adaptive • Manageable
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 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
PSTN
• 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?

Application
Integration:
Cisco IP Phones: PSTN Voice Mail
12SP+/30VIP Dialing Plans
Next Generation Supplementary
PBX Services PBX

Gateways
Provide Analog or Digital Access
and QoS support

WAN LAN Switches provide

QoS, inline Power and
Cisco Switched Infrastructure
CallManager

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 Open, Standards-Based

Application Legacy
Servers Cisco PBX
Directory
Server

PSTN

CallManager Gateway
QoS Enabled
Catalyst Switch
IP
Cisco IP Phones Voice-Enabled
Router

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 Open System Applications—
New World Ecosystem
Call Center Applications Voice Mail/UMS from
from GeoTel Acquisition Amteva Acquisition

Application
Servers

PSTN
Meet Me Conference

CallManager QoS Gateway

Enabled
Catalyst IP
Switch
Telekol

Picazo
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 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

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 …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

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 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
P
S
T
N

PBX

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 Cisco Multiservice Gateways

Headquarters
Scalabiliy and Density

DE 30+ Central Site/ 7513

Regional Branch
Enterprise 7507
7206
Branch
Small 7505
Branch 3660 7204
5300 7202
3640

2600 3620
3810
1750

Performance
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 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

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 Cisco AVVID WEB Attendant Console

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 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

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 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

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 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)

G.729(a)
.1 8k + Header 20k/10k (on PPP)

G.723.1 6.3k/5.3kk + Header 18k/8k (on PPP)

*Current Phones Use These Codecs

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 VoIP Low Speed Link (<768 KBPS)
Challenges and Solutions
Challenge Cisco Solutions
Congestion Intelligent Queuing
WFQ, IP Precedence, RSVP,
Delay and Delay Jitter
Priority Queuing

Packet Residency Interleaving

Slow Link Freeze-out by FRF.12, MLPPP, IP MTU Size Reduction,
Large Packets Faster Link

Bandwidth Consumption Compression

Header Size on Low Codecs, RTP Header Compression, Voice
Bandwidth Links Activity Detection

WAN Traffic Management

Router Traffic Shaping to CIR, High Priority
Oversubscription, Bursting
PVC, Data Discard Eligibility

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 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

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 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 Multilayer
Campus Campus

Campus WAN Edge WAN Backbone

Not as Critical “A Must” “A Must”
“Initially”
QoS Starts Often Overlooked
Must Be Switched in the WAN or Misunderstood

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 Policy Networking

• Define policies for applications

and users
• Distribute policy bindings
–QoS Policy Servers, Security
–Network enforcement nodes Telecommuters

• Enable integrated control over

network resources Campus
Mobile
Users

QoS
Security

Enterprise Policy Branch

Offices
Partners
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 IP Phone Roadmap
Remember IP Devices That Support Voice

Presentation_ID
F0_4339_c1 © 1999, Cisco Systems, Inc. 28
 Not Committed

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 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
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 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

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 Call Manager Release 3.0
Planned FCS Q2 CY 2000

Presentation_ID
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 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 (abrookes@cisco.com) or Eric Chapat (echapat@cisco.com)

• 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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 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

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 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

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 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

San Jose
–2500 Devices Per CM SanJoseE
Cluster SanJoseB
–512M Memory Required

• 10,000 IP Phones Per Cluster

–(5-1) * 2500 = 10,000 SanJoseD SanJoseC

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 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

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 How Do Clusters Communicate
Internally
Sequel Intra Cluster
Database Signaling = Full Mesh

7830
Publisher

7820 7820

Subscriber Subscriber

7820 7820

Subscriber Subscriber

Zermatt 2000 - Slide number 39 of 65

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 N+1 example

• Three devices are homed to

SanJoseD. All nodes in the network
are connected and are relaying route SanJoseA
and registration information to each
other.
SanJoseE SanJoseB

SanJoseD SanJoseC

Zermatt 2000 - Slide number 40 of 65

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 N+1 example

• SanJoseD Powered off. The devices lose their connection to CM SanJoseD.

SanJoseA

SanJoseE SanJoseB

X
SanJoseD SanJoseC

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 N+1 example
• The devices re-home to other call
managers, which then replicate new
route and registration information to SanJoseA
each other.
• The devices experience only a brief
outage - (TBD)
SanJoseE SanJoseB
–Calls in progress are dropped.
–Gateway calls are dropped.

• Since device operation is identical, X

SanJoseD SanJoseC
users may not notice that anything
happened.

Zermatt 2000 - Slide number 42 of 65

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 Call Manager Groups

• Each device -
–IP Phone
–Skinny Gateway

• Has a prioritized list of up to 3 Call Primary

Managers to which it can connect.
• This is called a CallManager Group Secondary

• This system is revertive (TBD)

Last Resort

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 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
• IP Prec = 5, DSCP = EF, Cos = 5 –Uses Pins 1,2,3,6

• 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

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 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

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 So How Does This All Work

Presentation_ID
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 Cisco AVVID : Basic Understanding

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 Cisco AVVID : IP to IP Call
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Zermatt 2000 - Slide number 49 of 65
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 Cisco AVVID : IP to PSTN Call
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 Cisco CallManager Voice Mail call
control 5201 dials 5202 - fwd to VM
Cisco All signaling and media over IP
Cisco uOne
CallManager 13-Message Waiting
Messaging Server

2-E.164 lookup 6-Call Setup

5050
7-Auto-Offhook
14-Message 8-Media 11-Greeting/tone
Waiting Connect playback (SMTP)
3-Call 10-Set
Setup Subscriber
1-Call 4-Alerting
11-Greeting/tone
Setup (Ring)
playback 9-Get
12-Record (SMTP) Subscriber
4-Alerting Message (RTP) 12-Record
(Ringback) Message
(SMTP)

5-No Answer
IMAP4 LDAP v3
Message Directory
Store

5202
5201
CFNA-->5050
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 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

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 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!

ngg
RRiin

Check My uOne
Calendar
ngg
RRiin

IP Network
Ring
i ng
ng R
Ri

Cell / PDA
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 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

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 Isolated Deployments
Amteva Gate Server Site B
Site A = Message Store
Directory
Call Call
Manager
Manager Router/GW
V V
IP WAN
Router
PSTN

Call
Call Manager at each site Manager
V
200 IP Phones
IP WAN
100 Voice Mail Boxes
Router
GW Selection based on PSTN signaling
–PRI - DT24+/DE-30+ or AS5300
–T1/E1 CAS - 2600/3600/7200 Site C
–MTP Required for IOS Gateways - Min 12.0(7)T

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 Multi-site WAN
“Distributed Call Processing”
Call
Site A Router/GW PSTN
Manager

V (Secondary V
Voice Path)
IP WAN
Router

IP WAN
Site B
(Primary Voice Path)

Call
Manager
Call Manager at Site
V
200 IP Phones IOS Gatekeeper for
Admission Control IP WAN
100 Voice Mail Boxes
Router
IOS Gateways Required
G.711/80kbps for ALL IP WAN voice calls
Maximum 10 IP WAN sites Site C
Transparent PSTN Fallback if WAN Unavailable
No Voice Mail Networking

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 Multi-site WAN
“Centralized Call Processing”
Centralized
Call Manager
Location 2
Location V
Router/GW
Hub PSTN
V
IP WAN
Router

IP WAN
Location 3
Centralized Dial Plan
V
200 IP Phones
100 Voice Mail Boxes
IP WAN
Admission Control - Location BW Limits
Router
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)

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 Multi-site WAN Topologies

Hub and Spoke Admission Control + Capacity Planning

Topologies
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
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 Voice Messaging
“All in one” Entry Package
MCS-7830
Messaging Directory
Server All components on
Voice/
Message Store/Directory Email
MCS-7830

uOne GateServer
Skinny Station Protocol
(4.1E)
Gateserver registers
Call Manager as an IP Phone
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”

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 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

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 3.0 Design Guide

Presentation_ID
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 CM 3.0 Isolated Deployments
A A

V V

PSTN
Site A
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
–MTP Not Required Site C
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 CM 3.0 Multi-Site Wan
Distributed Call processing
A A
Site A
PSTN
V V

Site B
IP WAN
(Primary Voice Path)
• Call Manager Cluster per site A

–10,000 IP Phones Per Cluster Gatekeeper

V
–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 Site C
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 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
Central Cluster1CMA 3. Creation of a “Locations Cluster” of two or three
4. Possible Location status synch during CM failover - self healing
Site
Primary Backup Backup
Cluster1CME Cluster1CMB
Central CM CM CM
Site A B C

Cluster1CMD Cluster1CMC
“Locations
WAN Cluster”

Remote
Remote
Sites
Sites
Location 1 Location 2
Location 1 Location 2

Not Supported Supported

Zermatt 2000 - Slide number 66 of 65
www.cisco.com
 3.0 Solution Set
No DSP Farm or Cisco CallManager at remote sites

Compressed Call Leg DSP = DSP Farm

G.711 Call Leg

Supported/Committed for 3.0

uOne Call
Gateserver 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 Router/GW
Site2
V PSTN V
GW

Dial “9” for Local PSTN GW

IP WAN
Dial “9” for Local PSTN GW

Partition Calling Search Space

1. Devices with similar “reachability 1. Which Partitions a device may search in for a
characteristics” dialed number
2. Items placed in Partition: 2. Provides dialing permissions/restrictions
IP Phones, DN’s, Gateways + Route 3. Each device “assigned” a Calling Search Space
Patterns
Zermatt 2000 - Slide number 68 of 65
www.cisco.com
 Dial Plan - Robust Digit Manipulation
337-1XXX Gatekeeper 447-2XXX
IP WAN
1111 2222

V V
PSTN
3600 “Voice 3600
User dials Call Manager Overflow” 2222 Presented
447-2222 strips “447” for IP WAN to Call Manager

Call Placed Across IP WAN

Call rejected Across IP WAN, takes PSTN
337-1XXX Gatekeeper 447-2XXX
IP WAN
1111 2222

V V
PSTN
3600 “Voice 3600
User dials Overflow” 2222 Presented
Call Manager
447-2222 to Call Manager
pre-pends “1610” for PSTN
Zermatt 2000 - Slide number 69 of 65
www.cisco.com
 MGCP
Elimination of MTP
Skinny MGCP Voice
Station Path

Call Manager Call Manager

with with
Amteva Gateserver Amteva Gateserver

V PSTN V PSTN

IOS GW 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 H323v2 Voice

Station Path

Cisco CallManager Cisco CallManager

with with
Amteva Gateserver Amteva Gateserver

V PSTN V PSTN

IOS GW 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. IOS gateways with Digital Interfaces without MTP
3. Minimum IOS 12.0(7)T

Zermatt 2000 - Slide number 71 of 65

www.cisco.com
 Call Manager and DSP Farm at each site
Compressed Call Leg
G.711 Call Leg
V = DSP Farm

Site A Site B
IOS Gatekeeper
CallManager Cisco uOne
Cisco uOne CallManager Cluster Gateserver
Gateserver Cluster

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
obridle@cisco.com

Presentation_ID © 1999, Cisco Systems, Inc. 75