byby

Wojciech Nawrot, Wojciech Śronek, and Krzysztof TurzaWojciech Nawrot, Wojciech Śronek, and Krzysztof Turza

Poznań 2005Poznań 2005

PBX REPLACEMENTPBX REPLACEMENT

Presentation planPresentation plan

Chapter 1. Chapter 1. PBX replacement stagesPBX replacement stages

Chapter 2. Chapter 2. CIPT – Cisco IP TelephonyCIPT – Cisco IP Telephony

Chapter 3. Chapter 3. VoIP signallingVoIP signalling

Chapter 4. Chapter 4. Quality of ServiceQuality of Service

Chapter 5. Chapter 5. Cisco IP Telephony deployment in a small companyCisco IP Telephony deployment in a small company

Chapter 6. Chapter 6. CIPT’s supplementary servicesCIPT’s supplementary services

Chapter 7.Chapter 7. Bibliography Bibliography

QuestionsQuestions

PBX replacement stagesPBX replacement stages

Chapter 1Chapter 1

IP WAN

PSTN

Telephone Network

Data Network

PBX

Telephone Network

Data Network

Office B

PBX

Office A

PBX replacement stages: traditional scenario PBX replacement stages: traditional scenario

• Two co-existing network architecturesTwo co-existing network architectures

• Separate links for voice and data between two sitesSeparate links for voice and data between two sites

• IP WAN as primary voice path (Long-distance voice traffic)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 processingPSTN as secondary (backup) voice path for traditional call processing

PSTN

PBX replacement stages: step 1 of 2 (integration)PBX replacement stages: step 1 of 2 (integration)

Voice Gateway Voice Gateway

IP WAN

Shared data & voice network

Shared data & voice network

Office BOffice A

PSTN

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

IP Phones IP Phones

• Modern IP phonesModern IP phones

Call ServerCall Server

Analog Phones Analog Phones

• Legacy analog phonesLegacy analog phones

Benefits of replacing existing PBX / PSTN systems with IP telephonyBenefits of replacing existing PBX / PSTN systems with IP telephony

• enhanced speech qualityenhanced speech quality

• G.722 – 7kHz speech bandwidthG.722 – 7kHz speech bandwidth

• improved coverage improved coverage

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

• cost reductioncost reduction

• free Internet calls between remote company branchesfree Internet calls between remote company branches

• cheap Internet worldwide calls by the agency of a carriercheap Internet worldwide calls by the agency of a carrier

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

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

• free conference connections eliminate dependence upon service providersfree conference connections eliminate dependence upon service providers

• low administration costs – in small companies no distinct technicans are necessary for separate voice low administration costs – in small companies no distinct technicans are necessary for separate voice and data and data

• the number of network service providers would be reducedthe number of network service providers would be reduced

• improved mobilityimproved mobility

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

• subscriber’s accessibility at the same Directory Number all over the worldsubscriber’s accessibility at the same Directory Number all over the world• new services & open standardsnew services & open standards

CIPT - Cisco IP TelephonyCIPT - Cisco IP Telephony

Chapter 2Chapter 2

Introduction to Cisco IP TelephonyIntroduction to Cisco IP Telephony

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

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

Analog Phone

Cisco IP Telephony components (1 of 3)Cisco IP Telephony components (1 of 3)

A

PSTN

CallManager Cluster

Cisco IP Phones IP Softphone

Router

VoIP WAN

Switch Gateway

• Cisco IP PhonesCisco IP Phones • feature-rich devicesfeature-rich devices

• contain DSPs for voice signal digitizingcontain DSPs for voice signal digitizing

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

• Cisco SoftphonesCisco Softphones• virtual phones that run in a Windows desktop PC or laptopvirtual phones that run in a Windows desktop PC or laptop

• the IP softphones digitize the voice signals and send the voice packets the IP softphones digitize the voice signals and send the voice packets across the IP network across the IP network

• the PCs contain speakers and microphones that can operate similarly the PCs contain speakers and microphones that can operate similarly to telephone handset to telephone handset

• softphones provide a rich environment for development of TAPI applicationssoftphones provide a rich environment for development of TAPI applications

Cisco IP Telephony components (2 of 3)Cisco IP Telephony components (2 of 3)

A

PSTN

CallManager Cluster

Cisco IP Phones IP Softphone

Router

VoIP WAN

Switch Gateway

• Cisco Call ManagerCisco Call Manager

• software call-processing application that runs on a Cisco Media software call-processing application that runs on a Cisco Media Convergence Server (MCS) Convergence Server (MCS)

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

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

such as transcoding and conferencing such as transcoding and conferencing

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

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

• call managers perform the functions traditionally performed by PBXscall managers perform the functions traditionally performed by PBXs

A

Analog Phone

Analog Phone

Cisco IP Telephony components (3 of 3)Cisco IP Telephony components (3 of 3)

A

PSTN

CallManager Cluster

Cisco IP Phones IP Softphone

Router

VoIP WAN

Switch Gateway

• GatewaysGateways• provide an interface between the IP telephony network and the PSTNprovide an interface between the IP telephony network and the PSTN

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

• pass calls from office IP phone to an analog phone and vice versapass 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)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 convert the digital voice packets into a TDM stream or analog signal and transmit the call through the PSTN PSTN

• SwitchesSwitches• support inline power to the IP phones support inline power to the IP phones

• support VLANs & QoSsupport VLANs & QoS

Distributed Call Processing vs. Centralized Call ProcessingDistributed Call Processing vs. Centralized Call Processing

A APSTN

Sec. voice path

IP WANPri. voice path

Gatekeeper (CAC)

CM Cluster CM Cluster

Site A Site B

• Distributed Call ProcessingDistributed Call Processing

• 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

• Centralized Call ProcessingCentralized Call Processing

APSTN

Sec. voice path

IP WANPri. voice path

CM Cluster

Site A Site B

ISDNbackup

• Cisco CallManager cluster at each location – confined to a single 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 subscribed

for voice traffic

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

• DSP resources for conferencing and WAN transcoding at each site

• voice mail, unified messaging and DSP resources available at central site only

CIPT’s important featuresCIPT’s important features

• TranscodingTranscoding

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

• important in conference calling when the participants are not using the same codec 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)allow for different compression levels for intra (G.711) and inter-region connections (G.729)

• CallManager clusteringCallManager clustering

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

• redundancy for backup call processing (2 servers) redundancy for backup call processing (2 servers)

• dedicated database publisher for making configuration changes and producing call detail dedicated database publisher for making configuration changes and producing call detail records (1 server) records (1 server)

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

• Call Admission Control (CAC)Call Admission Control (CAC)

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

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

• Call routingCall routing

• Route Patterns, Lists and Groups for handling the PSTN call routing if the primary IP WAN Route Patterns, Lists and Groups for handling the PSTN call routing if the primary IP WAN path is down or congested path is down or congested

Cisco IP phone physical connectivity and registration processCisco IP phone physical connectivity and registration process

Physical connectivity:Physical connectivity:

• some models of Cisco switches provide inline some models of Cisco switches provide inline power for IP phones power for IP phones

• a single port on the switch can be used to provide a single port on the switch can be used to provide

connectivity to both the Cisco IP phone and the connectivity to both the Cisco IP phone and the computer (the phone acts as a switch) computer (the phone acts as a switch)

DHCP server

Cisco IP phone

Cisco CallManager

+TFTP server

Registration process: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 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 switchthe 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 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. 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 CallManagersthe 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 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 file and runtime code necessary for the phone to operate. The IP phone receives a Directory Number (DN)(DN)• the IP phone is ready to make and receive callsthe IP phone is ready to make and receive calls

VoIP signallingVoIP signalling

Chapter 3Chapter 3

• H.323H.323 is an ITU-T recommendiation umbrella set of standards that definesis an ITU-T recommendiation umbrella set of standards that defines components, protocols, and procedures necessary to provide audio, video, components, protocols, and procedures necessary to provide audio, video, and data communications over and data communications over IP-basedIP-based networks networks

• H.323 protocol stackH.323 protocol stack• RAS (Registration, Administration, and Status)RAS (Registration, Administration, and Status) is used between endpoints and gatekeepers is used between endpoints and gatekeepers

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

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

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

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

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

ISO Protocol LayerISO Protocol Layer StandardStandard

PresentationPresentation G.711, G.729, G.729a, etcG.711, G.729, G.729a, etcSessionSession H.323, H.245, H.225, RTCPH.323, H.245, H.225, RTCPTransportTransport RTP, UDPRTP, UDPNetworkNetwork IP, RSVP, WFQIP, RSVP, WFQLinkLink FR, ATM, ETH, etcFR, 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 Control Audio/Video

H.323 overviewH.323 overview

H.323 componentsH.323 components

• H.323 GatekeepersH.323 Gatekeepers are considered to be „brains” are considered to be „brains” of H.323 network, and provide the following services: of H.323 network, and provide the following services:

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

H.323 Gatekeeper

• H.323 GatewaysH.323 Gateways provide a means for H.323 network to provide a means for H.323 network to communicate to other networks, most typicaly PSTN or PBX communicate to other networks, most typicaly PSTN or PBX systems. The GW functionality generally includes: systems. The GW functionality generally includes:

- translating protocols- translating protocols- converting information formats- converting information formats- transferring information - transferring information

ISDN H.320

PSTN H.324

SIP

H.323 Gateway

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

H.323 Terminal

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

H.323 MCU

IP WAN

PSTN

Zone A Zone B

FXOE&ME1/T1

CallManager /H.323 MCU

CallManager /H.323 MCU

H.323 Gateway

H.323 Gateway

H.323 Gatekeeper

(Zone A)

H.323 Gatekeeper

(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

SGCP for Cisco IP phones

SGCPfor Cisco IP phones

H.323 call stages and signalling flows H.323 call stages and signalling flows

H.323 call stagesH.323 call stages

1) discovery and registration (RAS)1) discovery and registration (RAS)

2) call setup (H.225)2) call setup (H.225)

3) call signalling flows 3) call signalling flows

4) media stream and media control flows4) media stream and media control flows

5) call termination (RAS)5) call termination (RAS)

Quality of ServiceQuality of Service

Chapter 4Chapter 4

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

• voice traffic requires: latency ( less than 150ms ), jitter ( a few ms ), packet loss ( far voice traffic requires: latency ( less than 150ms ), jitter ( a few ms ), packet loss ( far less than 1 percent ) less than 1 percent )

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

• layer 2 or layer 3layer 2 or layer 3 classification at the edge of the networkclassification 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 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- at layer 3 using the 3 bits of the DSCP field in the ToS byte of the IP header

• QoS mechanisms:QoS mechanisms:

• resource reservation (to make sure that VoIP call has the sufficient bandwidth resource reservation (to make sure that VoIP call has the sufficient bandwidth allocated before the conversation takes place ) allocated before the conversation takes place )

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

• CAC ( Call Admission Control ) to ensure that network resources are not CAC ( Call Admission Control ) to ensure that network resources are not oversubscribed. Calls that exceed the specified bandwidth are either rerouted using 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 an alternative route such as the PSTN, or busy tone is returned to the calling party

Quality of ServiceQuality of Service

Cisco IP Telephony deployment in Cisco IP Telephony deployment in a small companya small company

Chapter 5Chapter 5

LAB’s architectureLAB’s architecture

• Computer network architectureComputer network architecture

• 3 remote branches and 1 private network3 remote branches and 1 private network

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

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

• Cisco PIX as an internet gateway for all the company’s officesCisco 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 Cisco Catalyst 3550 in the central office as a traffic concentrator for voice and data

• IP telephony architectureIP telephony architecture

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

• applications and services on the same server machine as CCMapplications and services on the same server machine as CCM

• secondary backup call processing via PBX emulating the PSTNsecondary backup call processing via PBX emulating the PSTN

• 3 Cisco IP phones and 1 legacy analog phone in the central office3 Cisco IP phones and 1 legacy analog phone in the central office

• 1 Cisco IP phone and 1 analog phone in the fixed branch office1 Cisco IP phone and 1 analog phone in the fixed branch office

• 2 Cisco Aironet access points for a portable Wi-Fi Cisco IP telephone2 Cisco Aironet access points for a portable Wi-Fi Cisco IP telephone

• GateKeeper not necessary as all the IP phones registered to the same CallManagerGateKeeper not necessary as all the IP phones registered to the same CallManager• Voice portsVoice ports

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

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 VPN Concentrator

Cisco PIX NAT

C1760 Access Router

C1760 Access Router

IP Phone

IP Phone

IP Phone

IP PhoneAnalog Phone

Wi-Fi AP

Cisco Call Manager

Wi-Fi mobile IP Phone

AP Roaming

Cisco App.

Server

Wi-Fi AP

Catalyst 3550

Analog Phone

LAB’s components and logical topology LAB’s components and logical topology

Private networwork – behind NAT

PSTN

Serial (DTE)

(MCS 7815)

PBX

Cisco Call Manager

FXO

IP WAN /

VPN

VLAN routing

Application Server

FXO

Serial (DCE)

FXS

Fa

FXS FaFa

Fa

Fa Fa

Fa Fa

Fa Fa

VLAN Internet

VLAN Private (data)

VLAN Voice

VLAN CallManager

VLAN Trunk

VLAN configuration and physical inter-component connections

PSTN

IP WAN

FXSFXS FXO FXO

IP IP IP

IP IP

John SmithDN1: 1100DN2: 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: 1200DN2: 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-Conference 1016, 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 architectureDial plan architecture

6652920 6652921

PSTN

IP WAN /

VPNFXSFXS FXO FXO

IP IP IP

IP IP

John SmithDN1: 1100DN2: 1101

IP

Kate ColeDN: 1102

Steve EdwardsDN: 1103

Margaret YorkDN: 1140

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

Tom JonesDN1: 1200DN2: 1201

Kris KnightDN: 1220

Simple voice connectivity scenariosSimple voice connectivity scenarios

• Inter-office IP – to - IP call (John Smith to Tom Jones)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 - IP call (Kris Knight to Peter Hanks)

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

• IP – to IP – to - - PSTN call (Steve Edwards to TNC 2005 participient :)PSTN call (Steve Edwards to TNC 2005 participient :)

PSTN backupPSTN backup

PSTN

IP WAN / VPN

FXSFXS FXO FXO

IP

IP IP

Kate ColeDN: 1102

Margaret YorkDN: 1140

Tom JonesDN1: 1200DN2: 1201

Kris KnightDN: 1220

• IP WAN is down or congestedIP WAN is down or congested

• the IP phone at the remote office is losing IP connectivity with Cisco CallManager and the IP phone at the remote office is losing IP connectivity with Cisco CallManager and is getting unavailable. Only remote analog is getting unavailable. Only remote analog phones are staying operational.phones are staying operational.

• the PSTN is used as a backup path for voice connectionsthe 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 In the Centralized Call Processing scenario, IP backup is necessary to allow the remote IP phones coming back into operability phones coming back into operability

IP ISDN

Central Office Branch Office

CCM

CIPT’s supplementary servicesCIPT’s supplementary services

Chapter 6Chapter 6

Supplementary services overviewSupplementary services overview

• Selected CIPT’s features and servicesSelected CIPT’s features and services

• Software Conference BridgeSoftware Conference Bridge

• Call Pickup & Group Call PickupCall Pickup & Group Call Pickup

• Call ParkCall Park

• Extended services & Telephony applicationsExtended services & Telephony applications

• Auto AttendantAuto Attendant

• Integrated Contact DistributionIntegrated Contact Distribution

• Extension MobilityExtension Mobility

• Other CIPT’s features and servicesOther CIPT’s features and services

• Meet-MeMeet-Me conferences allow users to dial into a conference conferences allow users to dial into a conference

Software Conference Bridge @ Cisco CallManagerSoftware Conference Bridge @ Cisco CallManager

IP WAN / VPN

IP

IP IP

IP

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

• Ad-HocAd-Hoc conferences allow the conference controller to let only conferences allow the conference controller to let only certain participants into the conference certain participants into the conference

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

SoftwareMCU

DN: 1016

Conference ControllerConference Controller

John SmithDN1: 1100 Peter Hanks

DN3: 1302 (Cisco IP Communicator)Tom JonesDN1: 1200

SoftwareMCU

CCM

IP IP

IP

Kate ColeDN: 1102

Steve EdwardsDN: 1103

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

• Kate Cole is dialing Call Pickup Group number Kate Cole is dialing Call Pickup Group number 10151015 to pickup the call to pickup the call

• The incoming call is picked up by Kate ColeThe incoming call is picked up by Kate Cole

CCM

• Call Pickup allows you to answer a call that comes Call Pickup allows you to answer a call that comes in on a directory number other than your own. When in on a directory number other than your own. When you hear an incoming call ringing on another phone, you hear an incoming call ringing on another phone, you can redirect the call to your phone by using the you can redirect the call to your phone by using the call pickup feature. call pickup feature.

• there are two types of Call Pickup available on Cisco IP there are two types of Call Pickup available on Cisco IP phones: phones:

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

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

Call Pickup & Group Call PickupCall Pickup & Group Call Pickup

ROOM A ROOM B

Call Pickup Group

DN: 1015

IP IP

IP

Kate ColeDN: 1102

Steve EdwardsDN: 1103

• Steve Edwards is answering a call from Kate’s Cole IP phoneSteve 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 he has to check something on his computer to answer the question of the calling person. He is parking the call on number the call on number 102X102X and is coming back to his own room. and is coming back to his own room.

• he is unparking the call by choosing he is unparking the call by choosing 102X102X on his IP phone and is continuing the conversation on his IP phone and is continuing the conversation

CCM

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

• the Call Park feature works within a Cisco CallManager the Call Park feature works within a Cisco CallManager cluster as well as between clusters. cluster as well as between clusters.

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

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

Call ParkCall Park

ROOM A ROOM B

Call Park DN range:

1020-1029

Cisco IP Phone services (1 of 2)Cisco IP Phone services (1 of 2)

• additional services let regard an IP phone as a developed work tooladditional services let regard an IP phone as a developed work tool• examples:examples:

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

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

Web and database server

internetHTTP/XML

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

CallManager Web server

Cisco IP Phone services (2 of 2)Cisco IP Phone services (2 of 2)

Cisco IP Telephony applicationsCisco IP Telephony applications

• Telephony Application Programming Interface (TAPI)Telephony Application Programming Interface (TAPI)• interoperability across various computer platforms – Java TAPIinteroperability across various computer platforms – Java TAPI• CRA (Cutomer Response Applications) platform:CRA (Cutomer Response Applications) platform:

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

CallManager with

Cisco IP Telephony directory

CRA platform

CRA Editor

Softphone

• the caller knows the PSTN company’s number but doesn’t know extensionsthe caller knows the PSTN company’s number but doesn’t know extensions

• Cisco Auto Attendant allows callers to Cisco Auto Attendant allows callers to locate people in the organization locate people in the organization

• the software interacts with the caller and the software interacts with the caller and allows the caller to search for and to allows the caller to search for and to select the extension of the party he is select the extension of the party he is trying to reach trying to reach

• Auto Attendant provides the following Auto Attendant provides the following script: script:

• answer a callanswer a call• plays a user-configurable welcome promptplays a user-configurable welcome prompt• plays a main menu prompt that asks the plays a main menu prompt that asks the caller to perform one of three actions: caller to perform one of three actions:

- press „0” for the operator- press „0” for the operator- press „1” to enter an extension - press „1” to enter an extension

numbernumber- press „2” to spell by name- press „2” to spell by name

PSTNFXS FXO

IP IP

IP

Kate ColeDN: 1102

Steve EdwardsDN: 1103

Margaret YorkDN: 1140

Auto Attendant & Call TransferAuto Attendant & Call Transfer

Example: PSTN-to-Company dialExample: PSTN-to-Company dial

• the caller is calling the company and is pressing „0” for the operatorthe caller is calling the company and is pressing „0” for the operator

• the operator is transfering the call to appropriate personthe operator is transfering the call to appropriate person

CCM

Auto Attendant

Operator

Example 2: Other scenarios .....Example 2: Other scenarios .....

Press „0” for the

operator

6652920

Cisco IP Integrated Contact DistributionCisco IP Integrated Contact Distribution

• queues and distributes incoming calls queues and distributes incoming calls destinated for groups of Cisco destinated for groups of Cisco CallManager users (agents) CallManager users (agents)

• inteligent routing based on data gathered inteligent routing based on data gathered during connection time, skills of agents, during connection time, skills of agents, state of queues, time of the day, etc… state of queues, time of the day, etc…

• comfortable software for agents and comfortable software for agents and supervisors that manages incoming calls supervisors that manages incoming calls

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

maintenance) maintenance)

PSTN

Agent

CCM+ ICD

application

Steve EdwardsTechnical SupportDN: 1103

Example: PSTN-to-Company dialExample: PSTN-to-Company dial

• the caller knows the PSTN company’s number for technical supportthe caller knows the PSTN company’s number for technical support

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

• the application is transfering the call to the available agentthe application is transfering the call to the available agent

Extension MobilityExtension 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 individual With extension mobility, instead of assigning offices, and desks to individual employees, several different employees share office spaces on a rotational basis. employees, several different employees share office spaces on a rotational basis. This approach usually gets used in work environments in which employees do not This approach usually gets used in work environments in which employees do not routinely conduct business in the same place every day. routinely conduct business in the same place every day.

• The extension mobility feature allows users to configure Cisco IP Phones 7940 / 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 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, adopts the user individual profile information, including line numbers, speed dials, services links, and other user-specific properties of a phone. services links, and other user-specific properties of a phone.

Other CIPT’s features and servicesOther CIPT’s features and services

• Cisco uOne Voice MessagingCisco uOne Voice Messaging

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

• Music on Hold (MoH)Music on Hold (MoH)

• The integrated Music on Hold (MOH) feature alllows users to place on-net and 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 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 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, 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 serverPhone B listens to music that is streamed from a music on hold server

BibliographyBibliography

Chapter 7Chapter 7

BibliographyBibliography

• 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 ?Questions ?

Thank youThank you