cisco voip ppt

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

© 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

AgendaAgenda

• Why VoIP?

• Comparing & Understanding the VoIP Protocols- H.323

- Skinny- MGCP

- SIP

• SIP Tutorial

• Sample VoIP Applications

• Cisco VoIP products


Why VoIP? The Interesting StuffWhy 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


Why VoIP? The Challenging StuffWhy VoIP? The Challenging Stuff

• Do we need to replicate all the existing PSTN / PBX features?

• What’s 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


Open Packet TelephonyOpen Packet Telephony

TDM/TDM/Circuit SwitchCircuit Switch

Digital Trunk Digital Trunk SubsystemSubsystem

Digital Trunk Digital Trunk SubsystemSubsystem

Line Line ConcentrationConcentration

Line Line ConcentrationConcentration

AdministrationAdministrationMaintenanceMaintenance

BillingBilling

AdministrationAdministrationMaintenanceMaintenance

BillingBilling

Call ControlCall ControlConnection ControlConnection Control

FeaturesFeatures

Call ControlCall ControlConnection ControlConnection Control

FeaturesFeatures

Common ChannelCommon ChannelSignaling ComplexSignaling ComplexCommon ChannelCommon ChannelSignaling ComplexSignaling Complex

Sw

itch

ing

Net

wo

rk

Standards-Based Standards-Based Packet Infrastructure LayerPacket Infrastructure Layer

(IP, ATM)(IP, ATM)

Standards-Based Standards-Based Packet Infrastructure LayerPacket Infrastructure Layer

(IP, ATM)(IP, ATM)

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

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

Open Service Open Service Application LayerApplication Layer

(JAIN, AIN, TAPI,(JAIN, AIN, TAPI,JTAPI, XML etc.)JTAPI, XML etc.)

Open Service Open Service Application LayerApplication Layer

(JAIN, AIN, TAPI,(JAIN, AIN, TAPI,JTAPI, XML etc.)JTAPI, XML etc.)

Open/Standard Interface

Open/Standard Interface


The World Is Now Global—All Apps Must Travel Time and Distance

ApplicationsApplicationsApplicationsApplications

Call Call ProcessingProcessing

Call Call ProcessingProcessing

InfrastructureInfrastructureInfrastructureInfrastructure

ClientsClientsIP SoftPhone

• PSTN gateways• Analog phone support• DSP farms

IPNetwork

PSTN

Directory

Call Processing

Cisco UnityVoice Mail, UMS

Intelligent Contact Manager

IP IVR, IP AAApps Engine

Voice Portal

ICMICM

Collaboration

Video

AVVID Architecture - AVVID Architecture - Open Packet TelephonyOpen Packet Telephony

GKGK


AgendaAgenda

• Why VoIP?


- Skinny- MGCP

- SIP

• SIP Tutorial




VoIP Signaling ProtocolsVoIP 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 VoIP…SIMPLE: Instant Messaging / Presence


Gatekeeper A Gatekeeper B

RRQ/RCF

ARQ

RRQ/RCF

LRQ

IP Network

Phone A

Gateway A Gateway B

H.225 (Q.931) Setup

H.225 (Q.931) Alert and Connect

H.245

RTP

ACF

LCF

VV

Basic H.323 Call

VV

ARQ

ACF

Phone B


PSTN

CiscoCallManager

IP WAN

Voice Mail Server

Call Setup

E.164 Lookup

Ring

Off Hook

RTP Stream

Ring Back

Basic Skinny Call

H.323/MGCP Gateway


MGCP Architectures & Mixed ProtocolsMGCP Architectures & Mixed Protocols

PSTN

BTS / VSC

SS7

PSTN GatewayPSTN Gateway SIP or H.323 NetworkSIP or H.323 Network

V

AccessGateway

SCP

MGCP

V

SIPH.32

3IMT

PRI

RTP SIP / H.323

GKGK

V

PSTN


AgendaAgenda

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


- Skinny- MGCP

- SIP

• SIP Tutorial




Why are we talking about SIP?Why are we talking about SIP?

• Cisco has never met a protocol it didn’t like…. - Customers haven’t chosen 1 protocol to define VoIP

• SIP is a very Internet friendly protocol, and Cisco likes Internet friendly stuff…. - SIP reuses a lot of Internet protocols & formatting

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

• It’s 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 which concern 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 by commercial vendors

• SIP-T - inter Call Agent (MGC) protocol for carrying SS7 / ISUP messaging - basically maps ISUP messaging to a MIME attachment

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


SIP Basics - ArchitectureSIP Basics - Architecture

Legacy PBX

SIP User Agents (UA)

Application Services

eMail LDAP Oracle XML

SIPSIP

RTP(Media)

SIP

CPLCPL3pcc

PSTN

CAS or PRI

INTELLIGENT

SERVICES

SIP Proxy, Registrar & Redirect Servers


SIP Basics - Architectural ElementsSIP Basics - Architectural Elements

• Clients: SIP Phones, Softphones, Gateways, Media Gateway Controllers, PDAs, Robots - User Agent Client (UAC) / User Agent Server (UAS)

- Originate & Terminate SIP requests

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

• Servers: - Proxy Server- Redirect Server - Registrar Server


SIP Servers/Services (cont)SIP Servers/Services (cont)

SIP User Agents

Registrar RedirectLocationDatabase

SIP Proxy

SIP Servers/Services

REGISTER“Here I am”

INVITE“I want to talk to another UA

Proxied INVITE“I’ll handle it for

you”

“Where is this name/phone#?”

3xx Redirection“They moved,

try this address”

SIP User Agents

SIP-GW


SIP MethodsSIP 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 AddressingSIP Addressing

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

• Support for Fully-Qualified Domain Names (FQDNs) using sip: URLs - sip: “John Doe” <[email protected]>

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

• Support for mixed addresses - sip:[email protected]; user=phone sip:[email protected]

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


Basic SIP Call-FlowBasic 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


Basic SIP Functionality -Basic SIP Functionality -Call ForkingCall 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.

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.


Basic SIP Functionality -Basic SIP Functionality -Call RedirectionCall Redirection

LOCAL PSTN

Proxy / Redirect Server

Location Database

392-1234


“Where is sip:[email protected]?”

“You need to contact 4721111”

3xx MovedContact: sip:[email protected]


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

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


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

LOCAL PSTN

SIP Controller - 3pcc Application

INVITE sip:1234 w/o SDP

x1234

18x / 200 OK w/ SDP

INVIT

E sip

:919

4721

111

w/ SDP o

f SIP

Phone

18x

/ 200

OK w

/ SDP ACK w/ SDP of SIP Gateway

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.

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


AgendaAgenda



- Skinny- MGCP

- SIP

• SIP Tutorial




Web PagesWeb Pages EnterpriseDatabase

EnterpriseDatabase

Application Toolkit

External Services Packaged Solutions

IP IVR

Voice Portal

Auto Attendant

Telephony

Directory Access

Web Access

DB Access

LDAPLDAPNotification

ServerNotification

Server

QueuingQueuing

PagingPagingE-Mail

VXML servicesVXML

services

ICMICM

Notification Services

Queuing (ACD)

Personalized Apps

Customer Apps

Notification Services

Queuing (ACD)

Personalized Apps

Customer Apps

UnityUnity

Application Engine ArchitectureApplication Engine Architecture


IP Telephony Appliance

- Corporate directory integration via LDAP

- Web site integration via XML

- Personalized menu’s via softkeys

Extensible interface with IP services offers clear differentiation to PBX connected devices

IP Phone Display ApplicationsIP Phone Display Applications

*


Convergence:Presence ServicesConvergence:Presence Services

Managing your communicationsthrough web browsers, Instant Messagingand mobile devices


Remote AgentsRemote Agents

SoftPhone

IP PhonesPSTNPSTN

IP

Central Site

IAQServer

Branch AgentsBranch Agents

Distribution Groups with Queuing for Resources2 Types of Queues

RequestorServicer

Informal Agent Queuing (IAQ)


Web Attendant

• Ubiquitous access via a browser

• Extension look-up via LDAP

• Easy of use with drag and drop interface

• Benefits:

Eliminates specialized receptionist phones

Access via URL

• Included with Call Manager 3.0(tbd)


IP Intranet

Voice Portal Solution

• Extracts XML information from web page into IP IVR

• Benefit

Only one place to configure and maintain data

Consistency

Lower admin costsPress #1 to Hear

Stock Quote

IP IVR

Stock Quote

*


VoiceXML VoiceXML

PSTN

CiscoVoice Gateway

RTSPServer

VoiceXML in IOS: HTTPServer

ArchitecturalModel: VXML Interpreter Context

DocumentServer

Implementation Platform

VXML Interpreter


AgendaAgenda



- Skinny- MGCP

- SIP

• SIP Tutorial

• VoIP Applications



Cisco VoIP ProductsCisco 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, IP Contact 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


Questions?

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]

cisco voip ppt

Documents

sip user agents

ip phones

proxy server

sdp

web attendant

comparing

work

rights reserved