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PBX REPLACEMENT

by
Wojciech Nawrot, Wojciech ronek, and Krzysztof Turza

Pozna 2005

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

PBX replacement stages


Chapter 1

PBX replacement stages:


Telephone Network

traditional scenario
Telephone Network

PBX

PBX

PSTN

Data Network

Data Network

IP WAN

Office A
Two co-existing network architectures Separate links for voice and data between two sites

Office B

PBX replacement stages:


Telephone Network

step 1 of 2 (integration)
Telephone Network

PBX

PBX

Data Network

trunk

trunk

PSTN

Data Network

IP WAN
Voice Gateway Voice Gateway

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

Office B

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

PBX replacement stages:


Shared data & voice network
IP Phones

step 2 of 2 (complete PBX replacement)


Shared data & voice network
Analog Phones IP Phones

Analog Phones

Call Server Call Server

PSTN

IP WAN

Office A
Modern IP phones Legacy analog phones

Office B

Benefits of replacing existing PBX / PSTN systems with IP telephony


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

improved mobility
no need to deal with ports on the PBX and change dial numbers while moving an IP phone to another room subscribers accessibility at the same Directory Number all over the world

new services & open standards enhanced speech quality


G.722 7kHz speech bandwidth

CIPT - Cisco IP Telephony


Chapter 2

Introduction to Cisco IP Telephony

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

CIPT is the cornerstone of Cisco VoIP solutions and is fast replacing traditional PBXs

Cisco IP Telephony components (1 of 3)


VoIP WAN A Switch Gateway PSTN Router CallManager Cluster Cisco IP Phones IP Softphone Analog Phone

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 packets across the IP network the PCs contain speakers and microphones that can operate similarly to telephone handset softphones provide a rich environment for development of TAPI applications

Cisco IP Telephony components (2 of 3)


VoIP WAN A Switch Gateway PSTN Router CallManager Cluster Cisco IP Phones IP Softphone Analog Phone

Cisco Call Manager


software 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

Cisco IP Telephony components (3 of 3)


VoIP WAN A Switch Gateway PSTN Router CallManager Cluster Cisco IP Phones IP Softphone Analog Phone

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 the PSTN

Switches
support inline power to the IP phones support VLANs & QoS

Distributed Call Processing vs. Centralized Call Processing


Distributed Call Processing
A PSTN
Sec. voice path CM Cluster

Centralized Call Processing


A PSTN
Sec. voice path

CM Cluster

IP WAN
Pri. voice path

CM Cluster

IP WAN
Pri. voice path

Gatekeeper (CAC)

ISDN
backup Site B Site A Site B

Site A

a distributed Cisco CallManager network is not cost effective solution for extending IP telephony to small or medium-sized branch offices with less than 20 users Cisco CallManager cluster at each location confined to a single campus transparent use of PSTN if IP WAN is unavailable compressed calls supported Cisco IOS gatekeeper for Call Admission Control (CAC) DSP resources for conferencing and WAN transcoding at each site

a centralized Cisco CallManager solution reduces equipment and operational expense and is a cost effective solution for for sites with less then 20 users IP phones at remote sites do not have Cisco CallManager manual use of the PSTN if the IP WAN is fully subscribed for voice traffic compressed calls supported CAC based on bandwidth by location voice mail, unified messaging and DSP resources available at central site only dial backup is required for IP phone service across the WAN in case the IP WAN goes down

CIPTs important features


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

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)

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 locations construct, for Distributed Call Processing it can be implemented with H.323 Gatekeeper that can limit the 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

Cisco IP phone physical connectivity and registration process


Physical connectivity:
DHCP server Cisco CallManager + TFTP server

some models of Cisco switches provide inline power for IP phones a single port on the switch can be used to provide connectivity to both the Cisco IP phone and the computer (the phone acts as a switch)

Cisco IP phone

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 configuration file 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

VoIP signalling
Chapter 3

H.323 overview
H.323 is an ITU-T recommendiation umbrella set of standards that defines components, protocols, and procedures necessary to provide audio, video, and data communications over IP-based networks 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 between H.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)
Audio/Video Control Control

Control H.225 (Q.931) H.245

Data

Audio G.7XX

Video H.26X

ISO Protocol Layer


RTCP RAS

Standard
G.711, G.729, G.729a, etc H.323, H.245, H.225, RTCP RTP, UDP IP, RSVP, WFQ FR, ATM, ETH, etc

T.120 RTP

TCP IP

UDP

Presentation Session Transport Network Link

H.323 components
H.323 Endpoints (Terminals) provide the user-to-network interfaces for H.323 protocol (IP phones or videoconferencing terminals)

H.323 Gateways provide a means for H.323 network to communicate to other networks, most typicaly PSTN or PBX systems. The GW functionality generally includes:
- translating protocols - converting information formats - transferring information
H.323 Terminal

ISDN H.320 SIP

PSTN H.324

H.323 Gatekeepers are considered to be brains of 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.323 Gateway H.323 MCU H.323


Gatekeeper

H.323 MCUs (Multipoint Control Units) provide conference support for three or more endpoints

H.323 call stages and signalling flows


IP phone Analog phone PBX PBX Analog phone IP phone

PSTN
SGCP for Cisco IP phones FXO E&M E1/T1

FXO E&M E1/T1 H.225, H.245, RTCP, RTP


Direct dialing

SGCP for Cisco IP phones H.225, H.245, RTCP, RTP

H.225, H.245, RTCP, RTP

CallManager / H.323 MCU

RAS

H.323 RAS Gateway

IP WAN

H.323 Gateway RAS

RAS

CallManager / H.323 MCU

Zone A
H.323 Gatekeeper (Zone A) RAS

Zone B
H.323 Gatekeeper (Zone B)

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)

Quality of Service
Chapter 4

Quality of Service
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 is accomplished 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 each router decides if to respect this request or not ) CAC ( Call Admission Control ) to ensure that network resources are not oversubscribed. 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

Cisco IP Telephony deployment in a small company


Chapter 5

LABs architecture
Computer network architecture
3 remote branches and 1 private network

2 fixed officess with Cisco 1760 access routers connected through the internet with VPN tunnel
1 mobile office with software Cisco VPN Client, connected to the central office with Cisco 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)

LABs components and logical topology


Branch Office (Warsaw)

Central Office (Poznan) Location A


IP Phone Analog Phone

Location B

PSTN
C1760 Access Router

Analog Phone

IP Phone

IP Phone Catalyst 3550 IP Phone

Wi-Fi AP Wi-Fi AP

Cisco PIX NAT

C1760 Access Router VPN tunnel

IP WAN
VPN tunnel Cisco Call Manager Cisco VPN Cisco Concentrator App. Server

Branch Office Mobile Location C

AP Roaming

Wi-Fi mobile IP Phone Private networwork behind NAT

Public Hotspot

Cisco VPN Client Cisco Softphone

Cisco IP Communicator
Ms NetMeeting

VLAN configuration and physical inter-component connections

PSTN

PBX Fa Fa Fa FXS FXO Fa Fa Fa Fa Fa Fa VLAN routing Serial (DTE) FXO Serial (DCE) FXS Fa

IP WAN / VPN
Cisco Call Manager Application Server VLAN Internet VLAN Private (data) VLAN Voice VLAN CallManager VLAN Trunk (MCS 7815)

Dial plan architecture


Central Office (Poznan) LOCATION A John Smith DN1: 1100 DN2: 1101 Kate Cole DN: 1102 Steve Edwards DN: 1103 Branch Office MOBILE LOCATION C Branch Office (Warsaw) LOCATION B Tom Jones DN1: 1200 DN2: 1201 Peter Hanks DN1: 1300 (NetMeeting) DN2: 1301 (NetMeeting) DN3: 1302 (Cisco IP Communicator) DN4: 1303 (Cisco IP Softphone)

IP

IP

IP

IP

IP

6652920

IP WAN

6652921

IP

FXS Margaret York DN: 1140

FXO

FXO

FXS

PSTN
Central Office (Poznan)

Kris Knight DN: 1220

Branch Office (Warsaw)

Branch Mobile Office

NAME:
POSITION: PH. MODEL: EXTENTION: EXTERN LINE: SERVICES

John Smith
Chairman C 7920 1100, 1101

Kate Cole
Secretary C 7940 1102

Steve Edwards
C 7902 1103 6652920

Margaret York
Analog POTS 1140

Tom Jones
Chief of Staff C 7960 1200, 1201

Kris Knight
Analog POTS 1220 6652921

Peter Hanks
Ms NetMeeting
1300, 1301, 1302, 1303

Technical Support Sales Manager

Technical Support Sales Represent.

Meet-Me-Conference 1016, Call Pickup 1015, Call Park 102X, Auto Attendant 1000, Integrated Contact Distribution 1005,

Simple voice connectivity scenarios


John Smith DN1: 1100 DN2: 1101 Kate Cole DN: 1102 Steve Edwards DN: 1103

Tom Jones DN1: 1200 DN2: 1201

IP

IP

IP

IP

Peter Hanks DN1: 1300 (NetMeeting) DN2: 1301 (NetMeeting) DN3: 1302 (Cisco IP comm) DN4: 1303 (Cisco IP Softphone)

IP

IP WAN / VPN
FXO FXO

IP

FXS Margaret York DN: 1140

FXS

PSTN

Kris Knight DN: 1220

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 :)

PSTN backup
Kate Cole DN: 1102 Tom Jones DN1: 1200 DN2: 1201

IP ISDN
IP

CCM
IP

IP WAN / VPN
FXS FXO FXO

IP

FXS

Margaret York DN: 1140

PSTN
Central Office

Kris Knight DN: 1220

Branch Office

IP WAN is down or congested the IP phone at the remote office is losing IP connectivity with Cisco CallManager and is getting unavailable. Only remote analog phones 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

CIPTs supplementary services


Chapter 6

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

Software Conference Bridge @ Cisco CallManager


John Smith DN1: 1100

CCM Software MCU DN: 1016


Peter Hanks DN3: 1302 (Cisco IP Communicator) Tom Jones DN1: 1200

Conference Controller

IP

IP

IP

IP WAN / VPN
Meet-Me on Monday at 10.00 a.m. DN: 1016

IP

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


Meet-Me conferences allow users to dial into a conference

Ad-Hoc conferences allow the conference controller to let only


certain participants into the conference

Call Pickup & Group Call Pickup


Steve Edwards DN: 1103

Kate Cole DN: 1102

Call Pickup allows you to answer a call that comes in 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 the call pickup feature. there are two types of Call Pickup available on Cisco IP phones:
- Call Pickup allows users to pick up incoming calls within their own group. The appropriate call pickup group number is dialed automatically when a user activates this feature. - Group Call Pickup allows users to pick up incoming calls within their own group or in other groups. Users must dial the appropriate call pickup group number when using this feature.

ROOM A IP

ROOM B

IP

IP

CCM

Call Pickup Group DN: 1015

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

Call Park
Steve Edwards DN: 1103

Kate Cole DN: 1102

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 CallManager cluster as well as between clusters.

ROOM A IP

ROOM B

IP

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

IP

you can park only one call at each call park extension number

Call Park DN range: 1020-1029

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 parking the 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

Cisco IP Phone services (1 of 2)


CallManager Web and database server Web server

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

internet

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

Cisco IP Phone services (2 of 2)

Cisco IP Telephony applications


CRA platform CallManager with Cisco IP Telephony directory

CRA Editor

Softphone

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

Auto Attendant & Call Transfer


CCM Auto Attendant
Kate Cole DN: 1102 Steve Edwards DN: 1103

Cisco Auto Attendant allows callers to locate people in the organization the software interacts with the caller and allows the caller to search for and to select the extension of the party he is trying to reach

Operator

Press 0 for the IP operator

IP

Auto Attendant provides the following script:


IP 6652920 FXO

PSTN

answer a call plays a user-configurable welcome prompt plays a main menu prompt that asks the caller to perform one of three actions:
- press 0 for the operator - press 1 to enter an extension number - press 2 to spell by name

FXS Margaret York DN: 1140

Example 2: Other scenarios ..... Example: PSTN-to-Company dial the caller knows the PSTN companys number but doesnt know extensions the caller is calling the company and is pressing 0 for the operator the operator is transfering the call to appropriate person

Cisco IP Integrated Contact Distribution


CCM + ICD application
Steve Edwards Technical Support DN: 1103

Agent

queues and distributes incoming calls destinated for groups of Cisco CallManager users (agents) inteligent routing based on data gathered during connection time, skills of agents, state of queues, time of the day, etc comfortable software for agents and supervisors that manages incoming calls

PSTN

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

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

Extension Mobility
CCM
Kate Cole DN: 1102

IP

IP

IP WAN / VPN

IP

Central Office

Branch Office

With extension mobility, instead of assigning offices, and desks to individual employees, several different employees share office spaces on a rotational basis. This approach usually gets used in work environments in which employees do not routinely 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 phone adopts the user individual profile information, including line numbers, speed dials, services links, and other user-specific properties of a phone.

Other CIPTs features and services


Cisco uOne Voice Messaging The Cisco Unified Open Network Exchange (uOne) optional software, available as part of Cisco IP Telephony Solutions, provides voice messaging capability to users when they are unavailable to answer calls. The uOne software uses the Skinny Station protocol to communicate with Cisco CallManager

Music on Hold (MoH)


The integrated Music on Hold (MOH) feature alllows users to place on-net and off-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 to phone 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

Bibliography
Chapter 7

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 3.3 Cisco IP Telephony Solution Reference Network Design Cisco CallManager Document - Release 4.0 Cisco IP Telephony Network Design Guide www.cisco.com and www.google.pl websites

Questions ?

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