PBX Replacement

PBX REPLACEMENT
by
Wojciech Nawrot, Wojciech Śronek, and Krzysztof Turza
Poznań 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. CIPT’s supplementary services
Chapter 7. Bibliography
Questions
PBX replacement stages
Chapter 1
PBX replacement stages:
traditional scenario
Telephone Network
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:
step 1 of 2 (integration)
Telephone Network
Telephone Network
PBX
Data Network
PBX
trunk
trunk
PSTN
Data Network
IP WAN
Voice Gateway
Voice Gateway
Office A
Office B
• IP WAN as primary voice path (Long-distance voice traffic)
• PSTN as secondary (backup) voice path for traditional call processing
PBX replacement stages:
step 2 of 2 (complete PBX replacement)
Shared data & voice
network
IP Phones
Shared data & voice
network
Analog Phones
Analog Phones
IP 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
• subscriber’s 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
Switch Gateway
A
PSTN
Router
CallManager Cluster
Cisco IP Phones
Analog Phone
IP Softphone
• Cisco IP Phones
• featurefeature-rich devices
• contain DSPs for voice signal digitizing
• variety of models:
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
Switch Gateway
A
PSTN
Router
CallManager Cluster
Analog Phone
Cisco IP Phones
IP Softphone
• 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
Switch Gateway
A
PSTN
Router
CallManager Cluster
Cisco IP Phones
Analog Phone
IP Softphone
• Gateways
• provide an interface between the IP telephony network and the PSTN
• needed to allow calls between the VoIP locations,
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)
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
• Centralized Call Processing
A
A
CM Cluster
IP WAN
PSTN
Sec. voice path
Sec. voice path
CM Cluster
CM Cluster
IP WAN
Pri. voice path
Pri. voice path
Gatekeeper (CAC)
ISDN
backup
Site A
Site B
Site A
Site B
• 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
• a centralized Cisco CallManager solution reduces
equipment and operational expense and is a cost effective
solution for for sites with less then 20 users
• Cisco CallManager cluster at each location – confined to a
single campus
• IP phones at remote sites do not have Cisco CallManager
• transparent use of PSTN if IP WAN is unavailable
• manual use of the PSTN if the IP WAN is fully subscribed
for voice traffic
• compressed calls supported
• compressed calls supported
• Cisco IOS gatekeeper for Call Admission Control (CAC)
• CAC based on bandwidth by location
• DSP resources for conferencing and WAN transcoding at
each site
• 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
CIPT’s 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
Cisco IP phone
• 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)
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
(Registration,, Administration,
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 endend-toto-end network transport functions suitable for applications transmitting realreal-time data
• RTCP provides for reliable information transfer once the audio stream has been established
(media stream management)
management)
• Codecs define the degree of compression and decompression algorithms (G.711, G.723, G.729)
Control
Data
Video
G.7XX
H.225
(Q.931)
Audio
H.245
Audio/Video
Control Control
H.26X
T.120
RTCP
RTP
TCP
UDP
IP
ISO Protocol Layer
Standard
Presentation
Session
Transport
Network
Link
G.711, G.729, G.729a, etc
H.323, H.245, H.225, RTCP
RTP, UDP
IP, RSVP, WFQ
FR, ATM, ETH, etc
RAS
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 Gatekeepers are considered to be „brains”
of H.323 network, and provide the following services:
ISDN
H.320
H.323
Terminal
PSTN
H.324
SIP
H.323
Gateway
H.323
MCU
- address translation
- admission control
- bandwidth control and management
- zone managment
- call authorization
- call control signalling
- call management
• H.323 MCUs (Multipoint Control Units) provide conference
support for three or more endpoints
H.323
Gatekeeper
H.323 call stages and signalling flows
IP phone
PBX
Analog phone
PBX
IP phone
Analog phone
PSTN
SGCP
for Cisco IP phones
FXO
E&M
E1/T1
H.225, H.245, RTCP, RTP
FXO
E&M
E1/T1
H.225, H.245, RTCP, RTP
SGCP
for Cisco IP phones
H.225, H.245, RTCP, RTP
Direct dialing
CallManager /
H.323 MCU
RAS
H.323
RAS
Gateway
IP WAN
H.323
Gateway
RAS
Zone A
RAS
CallManager /
H.323 MCU
Zone B
H.323
Gatekeeper
(Zone A)
RAS
H.323
Gatekeeper
(Zone B)
H.323 call stages
1) discovery and registration (RAS)
4) media stream and media control flows
2) call setup (H.225)
5) call termination (RAS)
3) call signalling flows
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
LAB’s 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 company’s 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)
LAB’s components and logical topology
Branch Office (Warsaw)
Central Office (Poznan)
Location B
PSTN
Location A
IP Phone
Analog Phone
IP Phone
Analog Phone
IP Phone
Catalyst 3550
C1760 Access
Router
IP Phone
Wi-Fi AP
C1760 Access
Router
Cisco PIX
NAT
VPN tunnel
Wi-Fi AP
IP WAN
VPN tunnel
AP Roaming
Cisco Call
Manager
Branch Office
Mobile Location C
Cisco VPN
Cisco
Concentrator
App.
Server
Wi-Fi mobile
IP Phone
Public Hotspot
Cisco VPN Client
Cisco Softphone
Cisco IP Communicator
Private networwork – behind NAT
Ms NetMeeting
VLAN configuration and physical inter-component connections
PSTN
PBX
Fa
Fa
Fa
FXS Fa
FXS
FXO
FXO
Fa
Fa
Fa
Fa
Fa
Serial (DCE)
Serial (DTE)
Fa
VLAN routing
IP WAN /
VPN
Cisco Call
Manager
Application
Server
(MCS 7815)
VLAN Internet
VLAN CallManager
VLAN Private (data)
VLAN Trunk
VLAN Voice
Dial plan architecture
Central Office (Poznan) LOCATION A
John Smith
DN1: 1100
DN2: 1101
Kate Cole
DN: 1102
IP
IP
Branch Office MOBILE LOCATION C
Steve Edwards
DN: 1103
Tom Jones
DN1: 1200
DN2: 1201
IP
IP
IP
FXS
Peter Hanks
DN1: 1300 (NetMeeting)
DN2: 1301 (NetMeeting)
DN3: 1302 (Cisco IP Communicator)
DN4: 1303 (Cisco IP Softphone)
IP WAN
6652920
IP
FXO
NAME:
John Smith
Kate Cole
Steve Edwards
POSITION:
Chairman
Secretary
PH. MODEL:
C 7920
EXTENTION:
1100, 1101
Branch Office (Warsaw)
Branch Mobile Office
Tom Jones
Kris Knight
Technical Support Sales Manager
Chief of Staff
Technical Support Sales Represent.
C 7940
C 7902
Analog POTS
C 7960
Analog POTS
Ms NetMeeting
1102
1103
1140
1200, 1201
1220
1300, 1301, 1302, 1303
6652920
Margaret York
FXS
Kris Knight
DN: 1220
PSTN
Central Office (Poznan)
SERVICES
6652921
FXO
Margaret York
DN: 1140
EXTERN LINE:
Branch Office (Warsaw) LOCATION B
Peter Hanks
6652921
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
IP
Kate Cole
DN: 1102
IP
Steve Edwards
DN: 1103
IP
Margaret York
DN: 1140
IP
Peter Hanks
DN1: 1300 (NetMeeting)
DN2: 1301 (NetMeeting)
DN3: 1302 (Cisco IP comm)
DN4: 1303 (Cisco IP Softphone)
IP WAN /
VPN
IP
FXS
Tom Jones
DN1: 1200
DN2: 1201
IP
FXO
FXO
PSTN
• 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 :)
Kris Knight
DN: 1220
FXS
PSTN backup
Tom Jones
DN1: 1200
DN2: 1201
Kate Cole
DN: 1102
IP ISDN
IP
CCM
IP WAN /
VPN
IP
FXS
FXO
FXO
PSTN
Margaret York
DN: 1140
IP
Central Office
FXS
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
CIPT’s supplementary services
Chapter 6
Supplementary services overview
• Selected CIPT’s 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 CIPT’s features and services
Software Conference Bridge @ Cisco CallManager
CCM
John Smith
DN1: 1100
Peter Hanks
DN3: 1302 (Cisco IP Communicator)
Software
MCU
Conference
Controller
Tom Jones
DN1: 1200
DN: 1016
IP
IP
IP WAN /
VPN
IP
IP
Meet-Me on Monday at 10.00 a.m.
DN: 1016
• 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
ROOM B
ROOM A
IP
IP
IP
• 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:
CCM
- Call Pickup allows users to pick up incoming calls within their own
group.
group. The appropriate call pickup group number is dialed
automatically when a user activates this feature.
feature.
- Group Call Pickup allows users to pick up incoming calls within their
own group or in other groups.
groups. Users must dial the appropriate call
pickup group number when using this feature.
feature.
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 B
ROOM A
IP
• you can define either a single directory number or a
range of directory numbers for use as call park
extension numbers
IP
IP
CCM
• 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 Kate’s 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
HTTP/XML
Web server
internet
http://Web_Server/Stockquote.asp?stock=TPSA
• 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
• Cisco Auto Attendant allows callers to
locate people in the organization
CCM
Auto
Attendant
Kate Cole
DN: 1102
Steve Edwards
DN: 1103
Operator
Press „0”
for the IP
operator
• 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
IP
• Auto Attendant provides the following
script:
IP
6652920
FXS
PSTN
FXO
• answer a call
• plays a user-configurable welcome prompt
• plays a main menu prompt that asks the
caller to perform one of three actions:
Margaret York
DN: 1140
- press „0” for the operator
- press „1” to enter an extension number
- press „2” to spell by name
Example:2:PSTN
Other
-toscenarios
-Company.....
dial
• the caller knows the PSTN company’s number but doesn’t 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
• queues and distributes incoming calls
destinated for groups of Cisco
CallManager users (agents)
Agent
• 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 company’s 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
Central Office
IP WAN /
VPN
IP
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 CIPT’s 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 ?
Thank you