p Bx Replacement

7/31/2019 p Bx Replacement

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by

Wojciech Nawrot, Wojciech ronek, and Krzysztof Turza

Pozna 2005

PBX REPLACEMENT


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Presentation plan

Chapter 1. PBX replacement stages

Chapter 2. CIPT Cisco IP Telephony

Chapter 3. VoIP signalling

Chapter 4. Quality of Service

Chapter 5. Cisco IP Telephony deployment in a small company

Chapter 6. CIPTs supplementary services

Chapter 7. Bibliography

Questions


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PBX replacement stages

Chapter 1


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IP WAN

PSTN

Telephone Network

Data Network

PBX

Telephone Network

Data Network

Office B

PBX

Office A

PBX replacement stages: traditional scenario

Two co-existing network architectures

Separate links for voice and data between two sites


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IP WAN as primary voice path (Long-distance voice traffic)

IP WAN

Telephone Network

Data Network

PBX

Telephone Network

Data Network

PBX

trunk trunk

Office BOffice A

PSTN as secondary (backup) voice path for traditional call processing

PSTN

PBX replacement stages: step 1 of 2 (integration)

Voice Gateway Voice Gateway


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IP WAN

Shared data & voice

network

Shared data & voice

network

Office BOffice A

PSTN

PBX replacement stages: step 2 of 2 (complete PBX replacement)

IP Phones IP Phones

Modern IP phones

Call ServerCall Server

Analog Phones Analog Phones

Legacy analog phones


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Benefits of replacing existing PBX / PSTN systems with IP telephony

enhanced speech quality

G.722 7kHz speech bandwidth

improved coverage

in officess or laboratories offten a single phone is shared. Using workstation-based IPtelephony every employee is accessible at his own Directory Number

cost reduction

free Internet calls between remote company branches

cheap Internet worldwide calls by the agency of a carrier

no dedicated copper loops are necessary for an installation of new phones

free softphones can be used instead of hardphones (Ms NetMeeting)

free conference connections eliminate dependence upon service providers

low administration costs in small companies no distinct technicans are necessary for separate voice

and data the number of network service providers would be reduced

improved mobility

no need to deal with ports on the PBX and change dial numbers while moving an IP phoneto another room

subscribers accessibility at the same Directory Number all over the world

new services & open standards


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CIPT - Cisco IP Telephony

Chapter 2


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Introduction to Cisco IP Telephony

Cisco IP Telephony (CIPT) is the VoIP portion of the evolvingCisco Architecture for Voice, Video, and Integrated Data (AVVID)

CIPT is the cornerstone of Cisco VoIP solutions and is fastreplacing traditional PBXs


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Analog Phone

Cisco IP Telephony components (1 of 3)

A

PSTN

CallManager Cluster

Cisco IP Phones IP Softphone

Router

VoIP WAN

Switch Gateway

Cisco IP Phones

feature-rich devices

contain DSPs for voice signal digitizing

variety of models: 7960, 7940, 7920, 7912, 7902

Cisco Softphones

virtual phones that run in a Windows desktop PC or laptop

the IP softphones digitize the voice signals and send the voice packetsacross the IP network

the PCs contain speakers and microphones that can operate similarlyto telephone handset

softphones provide a rich environment for development of TAPI applications


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A

PSTN

CallManager Cluster


Router

VoIP WAN

Switch Gateway

Cisco Call Managersoftware call-processing application that runs on a Cisco Media

Convergence Server (MCS)

the CCM takes the place of a PBX and performs the following functions:

- registering IP Telephony devices, voice mail ports,TAPI & JTAPI devices, gateways and DSP resources

such as transcoding and conferencing

- call processing- administering dial plans and route plans- managing resources

a cluster of redundant CM groups can support up to 10k telephony users

call managers perform the functions traditionally performed by PBXs

A

Analog Phone


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Analog Phone


A

PSTN

CallManager Cluster


Router

VoIP WAN

Switch Gateway

Gateways

provide an interface between the IP telephony network and the PSTN

needed to allow calls between the VoIP locations, and PSTN locations

pass calls from office IP phone to an analog phone and vice versa

provide redundancy (divert outgoing calls from the WAN to the PSTN if the WAN is down or congested)

convert the digital voice packets into a TDM stream or analog signal and transmit the call through thePSTN

Switches

support inline power to the IP phones

support VLANs & QoS


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Distributed Call Processing vs. Centralized Call Processing

A APSTNSec. voice path

IP WANPri. voice path

Gatekeeper (CAC)

CM Cluster CM Cluster

Site A Site B

Distributed Call Processing

a distributed Cisco CallManager network is not costeffective solution for extending IP telephony to small ormedium-sized branch offices with less than 20 users

a centralized Cisco CallManager solutionreducesequipment and operational expense and is a cost effectivesolution for for sites with less then 20 users

Centralized Call Processing

A PSTNSec. voice path

IP WANPri. voice path

CM Cluster

Site A Site B

ISDNbackup

Cisco CallManager cluster at each location confined to asingle campus

IP phones at remote sites do not have Cisco CallManager

Cisco IOS gatekeeper for Call Admission Control (CAC) CAC based on bandwidth by location

compressed calls supported compressed calls supported

transparent use of PSTN if IP WAN is unavailable manual use of the PSTN if the IP WAN is fully subscribedfor voice traffic

dial backup is required for IP phone service across the WANin case the IP WAN goes down

DSP resources for conferencing and WAN transcoding ateach site

voice mail, unified messaging and DSP resources availableat central site only


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CIPTs important features

Transcoding

perform real-time translation of digitized voice from one codec to another

important in conference calling when the participants are not using the same codec

allow for different compression levels for intra (G.711) and inter-region connections (G.729)

CallManager clustering

increasing the system capacity (4 servers, 2500 IP phones per server)

redundancy for backup call processing (2 servers)

dedicated database publisher for making configuration changes and producing call detailrecords (1 server)

TFTP server for downloading of configuration files, device loads and ring types (1 server)

Call Admission Control (CAC)

a strategy used to limit the number of voice connections into the network in order to provide

the desired QoS

for Centralized Call Processing its provided using the locationsconstruct,for Distributed Call Processing it can be implemented with H.323 Gatekeeper that can limitthe maximum amount of bandwidth consumed by IP WAN voice calls in or out of the zone

Call routing

Route Patterns, Lists and Groups for handling the PSTN call routing if the primary IP WAN

path is down or congested


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Cisco IP phone physical connectivity and registration process

Physical connectivity:

some models of Cisco switches provide inlinepower for IP phones

a single port on the switch can be used to provideconnectivity to both the Cisco IP phone and thecomputer (the phone acts as a switch)

DHCP server

Cisco IP phone

CiscoCallManager

+TFTP server

Registration process:

the IP phone begins a CDP exchange with the switch and as a result it obtains VVID (Voice VLAN ID)

the IP phone issues a DHCP request on the voice subnet it got from the switch

the IP phone gets a response from the DHCP server. The response provides the IP address to the

telephone and the address of the TFTP server from which the phone gets its configuration.

the IP phone contacts the TFTP server and receives a list of addresses of Cisco CallManagers

the IP phone now contacts the Cisco CallManager and registers itself receiving in return a configurationfile and runtime code necessary for the phone to operate. The IP phone receives a Directory Number (DN)

the IP phone is ready to make and receive calls


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VoIP signalling

Chapter 3


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H.323is an ITU-T recommendiation umbrella set of standards that defines

components, protocols, and procedures necessary to provide audio, video,and data communications over IP-basednetworks

H.323 protocol stack

RAS (Registration, Administration, and Status) is used between endpoints and gatekeepers

H.225 (Q.931) provides call setup and control with all signalling necessary to establish a connection betweenH.323 endpoints

H.245 is used to negotiate channel usage and capabilities after setting up a call

RTP provides end-to-end network transport functions suitable for applications transmitting real-time data

RTCP provides for reliable information transfer once the audio stream has been established(media stream management)

Codecs define the degree of compression and decompression algorithms (G.711, G.723, G.729)

ISO Protocol Layer Standard

Presentation G.711, G.729, G.729a, etcSession H.323, H.245, H.225, RTCPTransport RTP, UDPNetwork IP, RSVP, WFQ

Link FR, ATM, ETH, etc

H.225

(Q.931) H.245 T.120 RTCP RAS

G.7XX H.26X

RTP

TCP UDP

IP

Control Data Audio Video Control ControlAudio/Video

H.323 overview


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H.323 components

H.323 Gatekeepersare considered to be brainsof H.323 network, and provide the following services:

- address translation

- admission control- bandwidth control and management- zone managment- call authorization- call control signalling- call management

H.323Gatekeeper

H.323 Gateways provide a means for H.323 network tocommunicate to other networks, most typicaly PSTN or PBXsystems. The GW functionality generally includes:

- translating protocols

- converting information formats

- transferring information

ISDNH.320

PSTNH.324

SIP

H.323Gateway

H.323 Endpoints (Terminals) provide the user-to-network interfaces for H.323

protocol (IP phones or videoconferencing terminals)

H.323Terminal

H.323 MCUs (Multipoint Control Units) provide conference

support for three or more endpoints

H.323MCU


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IP WAN

PSTN

Zone A Zone B

FXOE&ME1/T1

CallManager /

H.323 MCUCallManager /

H.323 MCUH.323Gateway

H.323Gateway

H.323Gatekeeper

(Zone A)

H.323Gatekeeper

(Zone B)

IP phonePBX PBX IP phone

RAS

RAS

RAS

H.225, H.245, RTCP, RTP

Direct dialing

RAS RAS

H.225, H.245, RTCP, RTPH.225, H.245, RTCP, RTP

Analog phoneAnalog phone

FXOE&ME1/T1

SGCPfor Cisco IP phones

SGCPfor Cisco IP phones

H.323 call stages and signalling flows

H.323 call stages

1) discovery and registration (RAS)

2) call setup (H.225)

3) call signalling flows

4) media stream and media control flows

5) call termination (RAS)


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Quality of Service

Chapter 4


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QoS refers to the capability of a network to provide better service to selected

network traffic voice traffic requires: latency ( less than 150ms ), jitter ( a few ms ), packet loss ( far

less than 1 percent )

the goal of protecting voice traffic from being run over by data traffic isaccomplished by classifying voice traffic as high priority

layer 2 or layer 3 classification at the edge of the network- at layer 2 using 3 bits in the 802.1p field which is a part of the 802.1q tag (CoS)

- at layer 3 using the 3 bits of the DSCP field in the ToS byte of the IP header

QoS mechanisms:

resource reservation (to make sure that VoIP call has the sufficient bandwidth

allocated before the conversation takes place )

traffic prioritization (the endpoint suggest a priority on the packets and eachrouter decides if to respect this request or not )

CAC ( Call Admission Control ) to ensure that network resources are notoversubscribed. Calls that exceed the specified bandwidth are either rerouted using

an alternative route such as the PSTN, or busy tone is returned to the calling party

Quality of Service


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Cisco IP Telephony deployment in

a small companyChapter 5


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LABs architecture

Computer network architecture

3 remote branches and 1 private network

2 fixed officess with Cisco 1760 access routers connected through the internet withVPN tunnel

1 mobile office with software Cisco VPN Client, connected to the central office withCisco VPN Concentrator

Cisco PIX as an internet gateway for all the companys offices

Cisco Catalyst 3550 in the central office as a traffic concentrator for voice and data

IP telephony architecture

centralized call processing model with a single Cisco CallManager server (MCS 7815)

applications and services on the same server machine as CCM

secondary backup call processing via PBX emulating the PSTN

3 Cisco IP phones and 1 legacy analog phone in the central office

1 Cisco IP phone and 1 analog phone in the fixed branch office

2 Cisco Aironet access points for a portable Wi-Fi Cisco IP telephone

GateKeeper not necessary as all the IP phones registered to the same CallManager

Voice ports

Every 1760 router with 2 VIC modules and 2 voice ports per module (FXS and FXO)


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Central Office (Poznan)

Location A

VPN tunnel

Branch Office (Warsaw)

Location B

Branch Office

Mobile Location C

IP WANVPN tunnel

PSTN

Public HotspotCisco VPN Client

Cisco Softphone

Cisco IP Communicator

Ms NetMeeting

Cisco VPNConcentrator

Cisco PIXNAT

C1760 AccessRouter

C1760 AccessRouter

IP Phone

IP Phone

IP Phone

IP PhoneAnalog Phone

Wi-Fi AP

Cisco Call

Manager

Wi-Fi mobileIP Phone

AP Roaming

Cisco

App.Server

Wi-Fi AP

Catalyst 3550

Analog Phone

LABs components and logical topology

Private networwork behind NAT


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PSTN

Serial (DTE)

(MCS 7815)

PBX

Cisco CallManager

FXO

IP WAN /

VPN

VLAN routing

ApplicationServer

FXO

Serial (DCE)

FXS

Fa

FXS FaFa

Fa

FaFa

Fa Fa

Fa Fa

VLAN Internet

VLAN Private (data)

VLAN Voice

VLAN CallManager

VLAN Trunk

VLAN configuration and physical inter-component connections


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PSTN

IP WAN

FXSFXS FXO FXO

IP IP IP

IP IP

John SmithDN1: 1100

DN2: 1101

IP

Kate ColeDN: 1102

Steve EdwardsDN: 1103

Margaret YorkDN: 1140

Peter HanksDN1: 1300 (NetMeeting)DN2: 1301 (NetMeeting)DN3: 1302 (Cisco IP Communicator)DN4: 1303 (Cisco IP Softphone)

Tom JonesDN1: 1200

DN2: 1201

Kris KnightDN: 1220

Central Office (Poznan) Branch Office (Warsaw) Branch Mobile Office

NAME: John Smith Kate Cole Steve Edwards Margaret York Tom Jones Kris Knight Peter Hanks

POSITION: Chairman Secretary Technical Support Sales Manager Chief of Staff Technical Support Sales Represent.

PH. MODEL: C 7920 C 7940 C 7902 Analog POTS C 7960 Analog POTS Ms NetMeeting

EXTENTION: 1100, 1101 1102 1103 1140 1200, 1201 1220 1300, 1301, 1302, 1303

EXTERN LINE: 6652920 6652921

SERVICES Meet-Me-Conference1016, Call Pickup 1015, Call Park 102X, Auto Attendant 1000, Integrated Contact Distribution 1005,

Central Office (Poznan) LOCATION A Branch Office MOBILE LOCATION C Branch Office (Warsaw) LOCATION B

Dial plan architecture

6652920 6652921


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PSTN

IP WAN /VPN

FXSFXS FXOFXO

IP IP IP

IP IP

John Smith

DN1: 1100DN2: 1101

IP

Kate Cole

DN: 1102

Steve Edwards

DN: 1103


Peter HanksDN1: 1300 (NetMeeting)DN2: 1301 (NetMeeting)DN3: 1302 (Cisco IP comm)DN4: 1303 (Cisco IP Softphone)

Tom Jones

DN1: 1200DN2: 1201

Kris KnightDN: 1220

Simple voice connectivity scenarios

Inter-office IP to - IP call (John Smith to Tom Jones)

Inter-office Analog to - IP call (Kris Knight to Peter Hanks)

Inter-office Analog to - Analog call (Kris Knight to Margaret York)

IP to-

PSTN call (Steve Edwards to TNC 2005 participient :)


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PSTN backup

PSTN

IP WAN /

VPN

FXSFXS FXOFXO

IP

IP IP

Kate ColeDN: 1102


Tom JonesDN1: 1200DN2: 1201

Kris KnightDN: 1220

IP WAN is down or congested

the IP phone at the remote office is losing IP connectivity with Cisco CallManager andis getting unavailable. Only remote analogphones are staying operational.

the PSTN is used as a backup path for voice connections

In the Centralized Call Processing scenario, IP backup is necessary to allow the remote IP

phones coming back into operability

IP ISDN

Central Office Branch Office

CCM


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CIPTs supplementary services

Chapter 6


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Supplementary services overview

Selected CIPTs features and services

Software Conference Bridge

Call Pickup & Group Call Pickup

Call Park

Extended services & Telephony applications

Auto Attendant

Integrated Contact Distribution

Extension Mobility

Other CIPTs features and services


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Meet-Me conferences allow users to dial into a conference

Software Conference Bridge @ Cisco CallManager

IP WAN /VPN

IP

IP IP

IP

Cisco CallManager supports both Meet-Me conferences and Ad-Hoc conferences:

Ad-Hoc conferences allow the conference controller to let onlycertain participants into the conference

Meet-Me on Monday at 10.00 a.m.DN: 1016

Software

MCU

DN: 1016

Conference

Controller

Conference

Controller

John SmithDN1: 1100

Peter HanksDN3: 1302 (Cisco IP Communicator)

Tom JonesDN1: 1200

Software

MCU

CCM


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IP IP

IP

Kate Cole

DN: 1102

Steve Edwards

DN: 1103

Steve Edwards is being called, but he is out of his room

Kate Cole is dialing Call Pickup Group number 1015 to pickup the call

The incoming call is picked up by Kate Cole

CCM

Call Pickup allows you to answer a call that comesin on a directory number other than your own. When

you hear an incoming call ringing on another phone,you can redirect the call to your phone by using thecall pickup feature.

there are two types of Call Pickup available on Cisco IPphones:

- Call Pickup allows users to pick up incoming calls within their owngroup. The appropriate call pickup group number is dialedautomatically when a user activates this feature.

- Group Call Pickup allows users to pick up incoming calls within theirown group or in other groups. Users must dial the appropriate callpickup group number when using this feature.

Call Pickup & Group Call Pickup

ROOM A ROOM B

Call Pickup Group

DN: 1015


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IP IP

IP

Kate Cole

DN: 1102

Steve Edwards

DN: 1103

Steve Edwards is answering a call from Kates Cole IP phone

he has to check something on his computer to answer the question of the calling person. He is parkingthe call on number 102X and is coming back to his own room.

he is unparking the call by choosing 102X on his IP phone and is continuing the conversation

CCM

the Call Park feature allows you to place a call on hold,so that it can be retrieved from another phone in the

system.

the Call Park feature works within a Cisco CallManagercluster as well as between clusters.

you can define either a single directory number or arange of directory numbers for use as call park

extension numbers

you can park only one call at each call park extensionnumber

Call Park

ROOM A ROOM B

Call Park DN range:

1020-1029


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Cisco IP Phone services (1 of 2)

additional services let regard an IP phone as a developed work tool

examples:

- personal address book- corporate directory- current stock value- business information about client

web applications (ASP/JSP) returns XML objects to the phone

Web and databaseserver

internetHTTP/XML

http://Web_Server/Stockquote.asp?stock=TPSA

CallManager Web server


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Cisco IP Phone services (2 of 2)


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Cisco IP Telephony applications

Telephony Application Programming Interface (TAPI)

interoperability across various computer platforms Java TAPI CRA (Cutomer Response Applications) platform:

- CRA application server with CRA Engine- CRA Editor and CRA administration web interface- application scripts are stored in LDAP directory- example of applications: auto attendant, integrated contact distribution

CallManagerwith

Cisco IPTelephonydirectory

CRA platform

CRA Editor

Softphone


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the caller knows the PSTN companys number but doesnt know extensions

Cisco Auto Attendant allows callers tolocate people in the organization

the software interacts with the caller andallows the caller to search for and toselect the extension of the party he istrying to reach

Auto Attendant provides the following

script: answer a call plays a user-configurable welcome prompt plays a main menu prompt that asks thecaller to perform one of three actions:

- press 0 for the operator- press 1 to enter an extension number- press 2 to spell by name

PSTNFXS FXO

IP IP

IP

Kate Cole

DN: 1102

Steve Edwards

DN: 1103


Auto Attendant & Call Transfer

Example: PSTN-to-Company dial

the caller is calling the company and is pressing 0 for the operator

the operator is transfering the call to appropriate person

CCM

Auto

AttendantOperator

Example 2: Other scenarios .....

Press 0for the

operator

6652920


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Cisco IP Integrated Contact Distribution

queues and distributes incoming calls

destinated for groups of CiscoCallManager users (agents)

inteligent routing based on data gatheredduring connection time, skills of agents,state of queues, time of the day, etc

comfortable software for agents andsupervisors that manages incoming calls

advantages (location independence,complete integration with CallManager,simplicity of installation, configuration and

maintenance)

PSTN

Agent

CCM+ ICD

application

Steve EdwardsTechnical SupportDN: 1103

Example: PSTN-to-Company dial

the caller knows the PSTN companys number for technical support

the caller is calling the company to gain a solution for his technical problem

the application is transfering the call to the available agent


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Extension Mobility

IP WAN /VPN

IP

IP IP

Kate ColeDN: 1102

CCM

Central Office Branch Office

With extension mobility, instead of assigning offices, and desks to individualemployees, several different employees share office spaces on a rotational basis.This approach usually gets used in work environments in which employees do notroutinely conduct business in the same place every day.

The extension mobility feature allows users to configure Cisco IP Phones 7940 /7960 as their own, by logging in to those phones. Once a user logs in, the phoneadopts the user individual profile information, including line numbers, speed dials,

services links, and other user-specific properties of a phone.


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Other CIPTs features and services

Cisco uOne Voice Messaging

The Cisco Unified Open Network Exchange (uOne) optional software, availableas part of Cisco IP Telephony Solutions, provides voice messaging capability tousers when they are unavailable to answer calls. The uOne software uses theSkinny Station protocol to communicate with Cisco CallManager

Music on Hold (MoH)

The integrated Music on Hold (MOH) feature alllows users to place on-net andoff-net users on hold with music that is streamed from a streaming source

In the simplest instance, music on hold takes effect when phone A is talking tophone B, and phone A places phone B on hold. If MOH resource is available

Phone B listens to music that is streamed from a music on hold server


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Bibliography

Chapter 7


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Bibliography

Margit Brandl, Dimitris Daskopoulos, Erik Dobbelsteijn, Jan Janak, Jiri Kuthan,

Saverio Niccolini, Jorg Ott, Stefan Prelle, Sven Ubik, Egon Verharen,IP Telephony Cookbook TERENA Report, March 2004

Robert Padjen, Larry Keefer, Sean Thurston, Jeff Bankston, Michael E. Flannagan,Martin Walshaw,Cisco AVVID and IP Telephony, Design & Implementation SYNGRESS

Paul J. Fong, Eric Knipp, David Gray, Scott M. Harris, Larry Keefer, Jr., Charles

Riley, Stuart Ruwet, Robert Thorstensen, Vincent Tillirson,Configuring Cisco, Voice over IP, SYNGRESS

Cisco CallManager Document - Release 3.3Cisco IP Telephony Solution Reference Network Design

Cisco CallManager Document - Release 4.0Cisco IP Telephony Network Design Guide

www.cisco.com and www.google.pl websites


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Questions ?


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Thank you

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