Voice over IP (VoIP)

gamgutturalMobile - Wireless

Dec 10, 2013 (3 years and 10 months ago)

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Internet2_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.

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© 2001, Cisco Systems, Inc. All rights reserved.

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Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

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Agenda


Why VoIP?


Comparing & Understanding the VoIP Protocols



-

H.323








-

Skinny








-

MGCP








-

SIP


SIP Tutorial


Sample VoIP Applications


Cisco VoIP products



© 2001, Cisco Systems, Inc. All rights reserved.

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© 2001, Cisco Systems, Inc. All rights reserved.

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Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

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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



© 2001, Cisco Systems, Inc. All rights reserved.

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© 2001, Cisco Systems, Inc. All rights reserved.

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Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

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Why 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

© 2001, Cisco Systems, Inc. All rights reserved.

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© 2001, Cisco Systems, Inc. All rights reserved.

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Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

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Open Packet Telephony

TDM/

Circuit Switch

Digital Trunk

Subsystem

Line

Concentration

Administration

Maintenance

Billing

Call Control

Connection Control

Features

Common Channel

Signaling Complex

Switching Network

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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Internet2_VoIP

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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

IP

Network

PSTN

Directory

Call

Processing

Cisco Unity

Voice Mail, UMS

Intelligent

Contact

Manager

IP IVR, IP AA

Apps Engine

Voice Portal

ICM

Collaboration

Video

AVVID Architecture
-


Open Packet Telephony

GK

© 2001, Cisco Systems, Inc. All rights reserved.

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© 2001, Cisco Systems, Inc. All rights reserved.

7


Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

7

Agenda


Why VoIP?


Comparing & Understanding the VoIP Protocols



-

H.323








-

Skinny








-

MGCP








-

SIP


SIP Tutorial


Sample VoIP Applications


Cisco VoIP products

© 2001, Cisco Systems, Inc. All rights reserved.

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© 2001, Cisco Systems, Inc. All rights reserved.

8


Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

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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 VoIP…SIMPLE: Instant Messaging / Presence

© 2001, Cisco Systems, Inc. All rights reserved.

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© 2001, Cisco Systems, Inc. All rights reserved.

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Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

9

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

V

Basic H.323 Call

V

ARQ

ACF

Phone B

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© 2001, Cisco Systems, Inc. All rights reserved.

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Internet2_VoIP

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PSTN

Cisco

CallManager

IP
WAN

Voice Mail Server

Call Setup

E.164 Lookup

Ring

Off Hook

RTP
Stream

Ring Back

Basic Skinny Call



H.323/MGCP
Gateway

© 2001, Cisco Systems, Inc. All rights reserved.

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© 2001, Cisco Systems, Inc. All rights reserved.

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Internet2_VoIP

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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

© 2001, Cisco Systems, Inc. All rights reserved.

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© 2001, Cisco Systems, Inc. All rights reserved.

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Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

12

Agenda


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


Comparing & Understanding the VoIP Protocols



-

H.323








-

Skinny








-

MGCP








-

SIP


SIP Tutorial


Sample VoIP Applications


Cisco VoIP products



© 2001, Cisco Systems, Inc. All rights reserved.

13

© 2001, Cisco Systems, Inc. All rights reserved.

13


Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

13

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


© 2001, Cisco Systems, Inc. All rights reserved.

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© 2001, Cisco Systems, Inc. All rights reserved.

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Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

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The history of SIP


S
ession
I
nitiation
P
rotocol (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)

© 2001, Cisco Systems, Inc. All rights reserved.

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© 2001, Cisco Systems, Inc. All rights reserved.

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Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

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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











© 2001, Cisco Systems, Inc. All rights reserved.

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Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

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SIP Basics
-

Architecture

Legacy PBX

SIP User

Agents (UA)

Application

Services

eMail


LDAP

Oracle

XML

SIP

SIP

RTP

(Media)

SIP

CPL

CPL

3pcc

PSTN

CAS or PRI

I

N
T
E
L
L

I

G
E
N
T


S
E
R
V

I

C
E
S

SIP Proxy, Registrar

& Redirect Servers

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Internet2_VoIP

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SIP 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









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Internet2_VoIP

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SIP Servers/Services (cont)

SIP User

Agents

Registrar

Redirect

Location
Database

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

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Internet2_VoIP

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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





© 2001, Cisco Systems, Inc. All rights reserved.

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© 2001, Cisco Systems, Inc. All rights reserved.

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Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

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SIP 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” <jdoe@cisco.com>


Support for E.164 addresses
-

sip:14085551234@gateway.com; user=phone


Support for mixed addresses
-





sip:14085551234@10.1.1.1; user=phone



sip:jdoe@10.1.1.1









Support for E.164 addresses using tel: URLs
-

tel:14085551234

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Internet2_VoIP

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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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Internet2_VoIP

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Basic SIP Functionality
-

Call Forking

LOCAL PSTN

Proxy / Redirect


Server

Location Database

INVITE sip:1
-
800
-
GO
-
CISCO@cisco.com

“Where is sip:1
-
800
-
GO
-
CISCO@cisco.com?”

“Contact 1234@10.1.1.1,

1234@10.1.1.2 and 1234@10.1.1.3”

INVITE sip:1234@10.1.1.1

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.

© 2001, Cisco Systems, Inc. All rights reserved.

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Internet2_VoIP

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Basic SIP Functionality
-

Call Redirection

LOCAL PSTN

Proxy / Redirect


Server

Location Database

392
-
1234

INVITE sip:3921234@cisco.com

“Where is sip:3921234@cisco.com?”

“You need to contact 4721111”

3xx Moved

Contact: sip:4721111@10.1.1.3

INVITE sip:4721111@10.1.1.3

National PSTN

The user at 392
-
1234 informed the network

that he could be reached on his cell
-
phone at 472
-
1111

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© 2001, Cisco Systems, Inc. All rights reserved.

24


Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

24

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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25

© 2001, Cisco Systems, Inc. All rights reserved.

25


Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

25

Agenda


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


Comparing & Understanding the VoIP Protocols



-

H.323








-

Skinny








-

MGCP








-

SIP


SIP Tutorial


Sample VoIP Applications


Cisco VoIP products



© 2001, Cisco Systems, Inc. All rights reserved.

26

© 2001, Cisco Systems, Inc. All rights reserved.

26


Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

26

Web Pages

Enterprise

Database

Application Toolkit

External

Services

Packaged Solutions

IP IVR

Voice Portal

Auto

Attendant

Telephony

Directory
Access

Web Access

DB Access

LDAP

Notification

Server

Queuing

Paging

E
-
Mail

VXML
services

ICM

Notification Services

Queuing (ACD)

Personalized Apps

Customer Apps

Unity

Application Engine Architecture


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© 2001, Cisco Systems, Inc. All rights reserved.

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Internet2_VoIP

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27

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 Applications

*

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© 2001, Cisco Systems, Inc. All rights reserved.

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Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

28

Convergence:Presence Services

Managing your communications

through web browsers, Instant Messaging

and mobile devices

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Internet2_VoIP

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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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Internet2_VoIP

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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)

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31

© 2001, Cisco Systems, Inc. All rights reserved.

31


Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

31

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 costs

Press #1 to Hear
Stock Quote

IP IVR

Stock Quote

*

© 2001, Cisco Systems, Inc. All rights reserved.

32

© 2001, Cisco Systems, Inc. All rights reserved.

32


Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

32

VoiceXML

PSTN

Cisco

Voice Gateway

RTSP

Server

VoiceXML in IOS:

HTTP

Server

Architectural

Model:

VXML Interpreter Context



Document

Server

Implementation Platform

VXML Interpreter

© 2001, Cisco Systems, Inc. All rights reserved.

33

© 2001, Cisco Systems, Inc. All rights reserved.

33


Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

33

Agenda


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


Comparing & Understanding the VoIP Protocols



-

H.323








-

Skinny








-

MGCP








-

SIP


SIP Tutorial


VoIP Applications


Cisco VoIP products



© 2001, Cisco Systems, Inc. All rights reserved.

34

© 2001, Cisco Systems, Inc. All rights reserved.

34


Internet2_VoIP

© 2001, Cisco Systems, Inc. All rights reserved.

34

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, 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


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Internet2_VoIP

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35

Questions?



36

Internet2_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
-

bgracely@cisco.com

37


Presentation_ID

© 2001, Cisco Systems, Inc. All rights reserved.