Cisco Ppt Voip

7/15/2019 Cisco Ppt Voip

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1Internet2_VoIP 2001, Cisco Systems, Inc. All rights reserved. 2001, Cisco Systems, Inc. All rights reserved. 2001, Cisco Systems, Inc. All rights reserved.

Voice over IP (VoIP)

Brian Gracely

Technical Marketing Engineer


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2001, Cisco Systems, Inc. All rights reserved. 2 2001, Cisco Systems, Inc. All rights reserved. 2Internet2_VoIP 2001, Cisco Systems, Inc. All rights reserved. 2

Agenda

Why VoIP?

Comparing & Understanding the VoIP Protocols- H.323- Skinny

- MGCP- SIP

SIP Tutorial

Sample VoIP Applications

Cisco VoIP products


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Why VoIP? The Interesting Stuff

Telecommunications Act of 1996- Deregulation of the Bell networks- Open the competitive markets for Service Providers

Converged Networks- Voice, Video & Data over an IP network- Reduced the costs of managing parallel networks- Allows voice to be an IP application

Centralized or distributed architectures

- Add features where they are needed


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Why VoIP? The Challenging Stuff

Do we need to replicate all the existing PSTN / PBXfeatures?

Whats the right architecture?

- Centralized- Distributed- Mix of both

How do we?- Provide better than PSTN QoS- Provide Admission Control- Secure the signaling & media- Meet all the regulatory requirements


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Open Packet Telephony

TDM/

Circuit Switch

Digital Trunk

Subsystem

LineConcentration

Administration

Maintenance

Billing

Call ControlConnection Control

Features

Common Channel

Signaling Complex

SwitchingNetwork

Standards-Based

Packet Infrastructure Layer(IP, ATM)

Open Call Control Layer(SIP, H.323, MGCP, etc.)

Open Service

Application Layer(JAIN, AIN, TAPI,

JTAPI, XML etc.)

Open/Standard

Interface

Open/Standard

Interface


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The World Is Now Global

All Apps Must Travel Time and Distance

Applications

Call

Processing

Infrastructure

Clients

IP SoftPhone

PSTN gateways

Analog phone support

DSP farms

IPNetwork

PSTN

Directory

Call

Processing

Cisco UnityVoice Mail, UMS

IntelligentContactManager

IP IVR, IP AAApps Engine

Voice Portal

ICM

Collaboration

Video

AVVID Architecture -Open Packet Telephony

GK


7/37 2001, Cisco Systems, Inc. All rights reserved. 7 2001, Cisco Systems, Inc. All rights reserved. 7Internet2_VoIP 2001, Cisco Systems, Inc. All rights reserved. 7

Agenda

Why VoIP?

Comparing & Understanding the VoIP Protocols- H.323- Skinny- MGCP- SIP

SIP Tutorial


Cisco VoIP products



VoIP Signaling Protocols

H.323- ITU standard, ISDN-based, distributed topology- 90%+ of all Service Provider VoIP networks- The current interconnect for CallManager to Service Providers- Useful for video applications

Skinny- Centralized Call-Control architecture.- CallManager controls all features.- over 700,000 IP Phones deployed

MGCP- IETF RFC2705

- Centralized Call-Control Architecture- Call-Agents (MGC) & Gateways (MG)

SIP- IETF RFC2543- Distributed Call-Control- Used for more than VoIPSIMPLE: Instant Messaging / Presence



Gatekeeper A Gatekeeper B

RRQ/RCF

ARQ

RRQ/RCF

LRQ

IP Network

Phone A

Gateway AGateway B

H.225 (Q.931) Setup

H.225 (Q.931) Alert and ConnectH.245

RTP

ACF

LCF

V

Basic H.323 Call

V

ARQ

ACF

Phone B



PSTN

Cisco

CallManagerIP

WAN

Voice Mail Server

Call Setup

E.164 Lookup

Ring

Off Hook

RTP

Stream

Ring Back

Basic Skinny Call

H.323/MGCPGateway



MGCP Architectures & Mixed Protocols

PSTN

BTS / VSC

SS7

PSTN Gateway SIP or H.323 Network

V

Access

Gateway

SCP

MGCP

V

SIP

H.323

IMT

PRI

RTP

SIP / H.323

GK

V

P

S

T

N



Agenda

Why VoIP? How does it work & why is it interesting?


- MGCP- SIP

SIP Tutorial


Cisco VoIP products



Why are we talking about SIP?

Cisco has never met a protocol it didnt like.- Customers havent chosen 1 protocol to define VoIP

SIP is a very Internet friendly protocol, and Cisco likes Internetfriendly stuff.

- SIP reuses a lot of Internet protocols & formatting

Customers still weary about proprietary protocols.- Skinny works well, but it is proprietary

Its about the Applications!!- The next Killer App is the integration of voice, data, video, IM &

Presence SIP can do this.

Microsoft!! 250 millions desktops might speak SIP soon. - SIP client will be added to WindowsXP in October



The history of SIP

Session Initiation Protocol (SIP) is defined via RFC2543 on March 17,1999.

Additional feature drafts have been written to address issues whichconcern SS7/ISUP handling, QoS, Alerting, DHCP, 3PCC, Firewalls &NAT, etc

IETF SIP-WG created in September, 1999

RFC2543bis (additions) created in April 2000.

Vendor interoperability testing done at the semi-annual SIP Bakeoff(8th in August in UK)



The various flavors of SIP

RFC2543 - vanilla SIP- the most commonly deployed & developed bycommercial vendors

SIP-T- inter Call Agent (MGC) protocol for carrying SS7 / ISUPmessaging

- basically maps ISUP messaging to a MIME attachment

SIP extension from PacketCable- additions to Security, QoS & Privacy areas



SIP Basics - Architecture

Legacy PBX

SIP UserAgents (UA)

Application

ServiceseMail LDAP Oracle XML

SIPSIP

RTP

(Media)

SIP

CPLCPL3pcc

PSTN

CAS or PRI

I

NT

E

L

L

I

GE

N

T

S

E

R

V

I

C

ES

SIP Proxy, Registrar

& Redirect Servers



SIP Basics - Architectural Elements

Clients: SIP Phones, Softphones, Gateways, MediaGateway Controllers, PDAs, Robots

- User Agent Client (UAC) / User Agent Server

(UAS)- Originate & Terminate SIP requests

Typically an endpoint will have both UAC & UAS, UAC fororiginating requests, and UAS for terminating requests

Servers:

- Proxy Server- Redirect Server- Registrar Server



SIP Servers/Services (cont)

SIP User

Agents

Registrar RedirectLocation

Database

SIP Proxy

SIP

Servers/

Services

REGISTER

Here I am

INVITE

I want to talkto another UA

Proxied INVITE

Ill handle it for

you

Where is this

name/phone#?

3xx RedirectionThey moved,

try this address

SIP User

AgentsSIP-GW



SIP Methods

Consists of Requests and Responses

Requests (unless mentioned, each has a response) REGISTER: UA registers with Registrar Server INVITE: request from a UAC to initiate a session ACK: confirms receipt of a final response to INVITE

BYE: sent by either side to end a call CANCEL: sent to end a call not yet connected OPTIONS: sent to query capabilities outside of SDP

Newly Adopted Methods: SUBSCRIBE & NOTIFY: used to identify device status /

presence. The foundation of SIP IM / Presence (IMPP). INFO: a means of carrying data in a message body REFER: the mechanism to initiate a Transfer MESSAGE: the means of carrying data for SIP IMPP

Messages contain SIP Headers and Body. Body might be SDP

or an attachment or some other application



SIP Addressing

Modeled after mailto URLs. May be a combination of FQDNs orE.164 numbers or both.

Support for Fully-Qualified Domain Names (FQDNs) using sip:

URLs - sip: John Doe

Support for E.164 addresses -sip:[email protected]; user=phone

Support for mixed addresses -

sip:[email protected]; user=phonesip:[email protected]

Support for E.164 addresses using tel: URLs - tel:14085551234


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Basic SIP Call-Flow

SIP UA1 SIP UA2

INVITE w/ SDP for Media Negotiation

100 Trying

180/183 Ringing w/ SDP for Media Negotiation

200 OK

200 OK

BYE

MEDIA

MEDIA

ACK


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Basic SIP Functionality -Call Forking

LOCAL PSTN

Proxy / Redirect

Server

Location Database

INVITE sip:[email protected]

Where is sip:[email protected]?

Contact [email protected],

[email protected] and [email protected]


Forked Calls can be in parallel or sequential. The first phone to

answer will get the call, the others will get a CANCEL from the

Proxy Server.


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Basic SIP Functionality -Call Redirection

LOCAL PSTN

Proxy / Redirect

Server

Location Database

392-1234


Where is sip:[email protected]?

You need to contact 4721111

3xx Moved

Contact: sip:[email protected]


National PSTNThe user at 392-1234 informed the networkthat he could be reached on his cell-phone at 472-1111


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3rd-Party Call-Control (3pcc) &Back-to-Back UserAgent (B2BUA)

LOCAL PSTN

SIP Controller - 3pcc Application

x1234

A user could manage their communications via a webpage. The webpage would invoke the SIP 3PCC

application to create SIP sessions to all parties involved.

HTTP post


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Agenda



- MGCP- SIP

SIP Tutorial


Cisco VoIP products


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Web Pages EnterpriseDatabase

Application Toolkit

External

ServicesPackaged Solutions

IP IVR

Voice Portal

AutoAttendant

Telephony

DirectoryAccess

Web Access

DB Access

LDAPNotification

Server

Queuing

PagingE-Mail

VXMLservices

ICM

Notification Services

Queuing (ACD)

Personalized Apps

Customer Apps

Unity

Application Engine Architecture


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IP Telephony Appliance

- Corporate directoryintegration via LDAP

- Web site integration viaXML

- Personalized menus viasoftkeys

Extensible interface withIP services offers cleardifferentiation to PBXconnected devices

IP Phone Display Applications

*


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Convergence:Presence Services

Managing your communications

through web browsers, Instant Messaging

and mobile devices


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Remote

Agents

SoftPhone

IP Phones

PSTN

IP

Central Site

IAQ

Server

Branch

Agents

Distribution Groups

with Queuing for

Resources

2 Types of Queues

Requestor

Servicer

Informal Agent Queuing (IAQ)


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Web Attendant

Ubiquitous access via abrowser

Extension look-up via LDAP

Easy of use with drag anddrop interface

Benefits:

Eliminates specializedreceptionist phones

Access via URL

Included withCall Manager 3.0(tbd)


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IP Intranet

Voice Portal Solution

Extracts XML informationfrom web page into IP IVR

Benefit

Only one place to configureand maintain data

Consistency

Lower admin costs

Press #1 to Hear

Stock Quote

IP IVR

Stock Quote

*


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VoiceXML

PSTN

CiscoVoice Gateway

RTSPServer

VoiceXML in IOS:HTTPServer

ArchitecturalModel: VXML Interpreter Context

Document

Server

Implementation Platform

VXML Interpreter


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Agenda



- MGCP- SIP

SIP Tutorial

VoIP Applications

Cisco VoIP products


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Cisco VoIP Products Call-Processing

- Cisco CallManager- Multimedia Conference Mgr - H.323 Gatekeeper / Proxy- Cisco SIP Proxy Server (CSPS)- BTS10200 Softswitch- VSC3000 Softswitch

VoIP Gateways

- Low End: ATA 186, 827v4, CVA122, uBR924, 1750, VG200- Mid Range: 3810, 2421, 2600, 3600, Cat4000, AS5300, 7200, 7500- High End: AS5350, AS5400, Cat6000, AS5850, MGX8850

IP Phones- 7910, 7940, 7960, 7935, Softphone

Applications- Unity UM, Personal Assistant, Conference Connection, IP IVR, IPContact Center, Web Attendant, XML / BTXML on IP Phones

- 80+ EcoSystem partners

Cisco Infrastructure- IOS QoS features, Line-Powered Catalyst Switches, Catalyst QoS

features- Application Layer Gateway (ALG) in IOS-NAT / Firewall, PIX


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Questions?


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36Internet2_VoIP 2001, Cisco Systems, Inc. All rights reserved. 2001, Cisco Systems, Inc. All rights reserved. 2001, Cisco Systems, Inc. All rights reserved.

Voice over IP (VoIP)

Brian Gracely - [email protected]


37/37

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