avvid update

8/8/2019 Avvid Update

1/75

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


2/75


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



3/75

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


4/75


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


5/75


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


6/75


7/75


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


8/75


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


9/75


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


10/75


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


11/75


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


12/75


The ComponentsFor Call Manager Release 2.4

The ComponentsFor Call Manager Release 2.4

12F0_4339_c1 1999, Cisco Systems, Inc.


13/75

www

.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


14/75

www


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


15/75

www


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


16/75

www


53005300 72027202

72047204

72067206

75057505

75077507

75137513

17501750

Cisco Multiservice GatewaysCisco Multiservice Gateways

Performance

2600260036203620

36403640

36603660

38103810


17/75

www


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


18/75

www


Cisco AVVID WEB Attendant ConsoleCisco AVVID WEB Attendant Console


19/75

www


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


20/75


QoS For IP TelephonyWithout it nothing works !

QoS For IP TelephonyWithout it nothing works !



21/75


22/75

www

.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


23/75

www

.cisco.com


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 (


24/75


Some RecommendationsSome Recommendations

24

F0_4339_c1 1999, Cisco Systems, Inc.


25/75

www

.cisco.com


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


26/75

www

.cisco.com


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


27/75

www

.cisco.com


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


28/75


IP Phone RoadmapRemember IP Devices That Support Voice

IP Phone RoadmapRemember IP Devices That Support Voice



29/75

www

.cisco.com


Not Committed


30/75

www

.cisco.com


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


31/75

www

.cisco.com


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


32/75


Call Manager Release 3.0Planned FCS Q2 CY 2000

Call Manager Release 3.0Planned FCS Q2 CY 2000



33/75

www

.cisco.com


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


34/75

www

.cisco.com


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


35/75


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


36/75


Distributed Call ProcessingDistributed Call Processing



37/75


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


38/75


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


39/75


How Do Clusters CommunicateInternally

How Do Clusters CommunicateInternally

Sequel

Database

Intra Cluster

Signaling = Full Mesh

Publisher

Subscriber

Subscriber Subscriber

Subscriber

7830

7820

78207820

7820


40/75


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


41/75



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

SanJoseA

SanJoseE

SanJoseD SanJoseC

SanJoseB

X


42/75


SanJoseA

SanJoseE

SanJoseD SanJoseC

SanJoseB

X


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.


43/75


44/75


Catalyst EnhancementsCatalyst Enhancements



45/75


46/75


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


47/75


So How Does This All WorkSo How Does This All Work



48/75


Cisco AVVID : Basic UnderstandingCisco AVVID : Basic Understanding


49/75


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


50/75


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


51/75


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


52/75


Application IntegrationApplication Integration



53/75


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


54/75


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


55/75


2.4 Design Guide2.4 Design Guide



56/75


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


57/75


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


58/75


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


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


59/75


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


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)


60/75


61/75


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


62/75


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


63/75


3.0 Design Guide3.0 Design Guide



64/75


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


2500 per Cal Manager

No Limit to Number of Clusters (Each Cluster is isolated PBX)

G.711 - Makes no sense to compress calls


MTP Not Required


65/75


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


Max 10 Clusters (5 Redundant) - IOS GK for CAC

G.711 or G.729 for WAN (C6K DSP Farm required for G.729 & Conference)


MTP Not Required


66/75


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


67/75


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


68/75


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


69/75


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


70/75


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


71/75


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


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


72/75


IOS Gatekeeper

V VRouter/GW Router/GW

CallManager

Cluster

CallManager

Cluster

IP WAN


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


73/75


74/75


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


75/75

75P t ti ID

Thank-you.Thank-you.

Owen Bridle

Cisco Voice & MultiService Technologies Consulting Engineer

Enterprise LOB

[email protected]

Owen Bridle

Cisco Voice & MultiService Technologies Consulting Engineer

Enterprise LOB

[email protected]

avvid update

Documents