PBX REPLACEMENT. Poznań and Krzysztof Turza. Wojciech Nawrot, Wojciech Śronek,

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1 PBX REPLACEMENT by Wojciech Nawrot, Wojciech Śronek, and Krzysztof Turza Poznań 2005

2 Presentation plan Chapter 1. PBX replacement stages Chapter 2. CT Cisco Telephony Chapter 3. Vo signalling Chapter 4. Quality of Service Chapter 5. Cisco Telephony deployment in a small company Chapter 6. CT s supplementary services Chapter 7. Bibliography Questions

3 PBX replacement stages Chapter 1

4 PBX replacement stages: traditional scenario Telephone Network Telephone Network PBX PBX PSTN Data Network Data Network WAN Office A Office B Two co-existing network architectures Separate links for voice and data between two sites

5 PBX replacement stages: : step 1 of 2 (integration( integration) Telephone Network Telephone Network PBX PBX Data Network trunk PSTN trunk Data Network WAN Voice Gateway Voice Gateway Office A Office B WAN as primary voice path (Long-distance voice traffic) PSTN as secondary (backup) voice path for traditional call processing

6 PBX replacement stages: : step 2 of 2 (complete( PBX replacement) Shared data & voice network Shared data & voice network Phones Analog Phones Analog Phones Phones Call Server Call Server PSTN WAN Office A Office B Modern phones Legacy analog phones

7 Benefits of replacing existing PBX / PSTN systems with 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 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 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

8 CT - Cisco Telephony Chapter 2

9 Introduction to Cisco Telephony Cisco Telephony (CT) is the Vo portion of the evolving Cisco Architecture for Voice,, Video, and Integrated Data (AVVID) CT is the cornerstone of Cisco Vo solutions and is fast replacing traditional PBXs

10 Cisco Telephony components (1 of 3) Vo WAN A Switch Gateway PSTN Router CallManager Cluster Analog Phone Cisco Phones Softphone Cisco Phones feature-rich 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 softphones digitize the voice signals and send the voice packets across the network the PCs contain speakers and microphones that can operate similarly to telephone handset softphones provide a rich environment for development of TAPI applications

11 Cisco Telephony components (2 of 3) Vo WAN A Switch Gateway PSTN Router CallManager Cluster Analog Phone Cisco Phones Cisco Call Manager Softphone 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 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

12 Cisco Telephony components (3 of 3) Vo WAN A Switch Gateway PSTN Router CallManager Cluster Analog Phone Cisco Phones Softphone Gateways provide an interface between the telephony network and the PSTN needed to allow calls between the Vo locations, and PSTN locations pass calls from office 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 phones support VLANs & QoS

13 Distributed Call Processing vs. Centralized Call Processing Distributed Call Processing Centralized Call Processing A PSTN Sec. voice path A A PSTN Sec. voice path CM Cluster WAN Pri. voice path CM Cluster CM Cluster WAN Pri. voice path Site A Gatekeeper (CAC) Site B Site A ISDN backup Site B a distributed Cisco CallManager network is not cost effective solution for extending 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 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 phones at remote sites do not have Cisco CallManager manual use of the PSTN if the 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 phone service across the WAN in case the WAN goes down

14 CT s important features CallManager clustering increasing the system capacity (4 servers,, 2500 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 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 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 WAN path is down or congested

15 Cisco phone physical connectivity and registration process Physical connectivity: Cisco phone DHCP server Cisco CallManager + TFTP server some models of Cisco switches provide inline power for phones a single port on the switch can be used to provide connectivity to both the Cisco phone and the computer (the phone acts as a switch) Registration process: the phone begins a CDP exchange with the switch and as a result it obtains VVID (Voice( VLAN ID) the phone issues a DHCP request on the voice subnet it got from the switch the phone gets a response from the DHCP server. The response provides the address to the telephone and the address of the TFTP server from which the phone gets its configuration. the phone contacts the TFTP server and receives a list of addresses of Cisco CallManagers the 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 phone receives a Directory Number (DN) the phone is ready to make and receive calls

16 Vo signalling Chapter 3

17 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 -based networks H.323 protocol stack RAS (Registration( 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 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) H.225 Audio/Video Control Data Audio Video Control Control (Q.931) H.245 T.120 RTCP RAS TCP G.7XX RTP H.26X UDP ISO Protocol Layer Presentation Session Transport Network Link Standard G.711, G.729, G.729a, etc H.323, H.245, H.225, RTCP RTP, UDP, RSVP, WFQ FR, ATM, ETH, etc

18 H.323 components H.323 Endpoints (Terminals) provide the user-to to-network interfaces for H.323 protocol ( 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: - 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 Terminal H.323 MCU ISDN H.320 S PSTN H.324 H.323 Gateway H.323 Gatekeeper

19 H.323 call stages and signalling flows phone Analog phone PBX PSTN PBX Analog phone phone CallManager / H.323 MCU Zone A SGCP for Cisco phones H.225, H.245, RTCP, RTP RAS FXO E&M E1/T1 H.323 Gateway H.225, H.245, RTCP, RTP RAS H.323 Gatekeeper (Zone A) Direct dialing WAN RAS H.323 Gatekeeper (Zone B) FXO E&M E1/T1 H.323 Gateway RAS SGCP for Cisco phones H.225, H.245, RTCP, RTP RAS CallManager / H.323 MCU 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)

20 Quality of Service Chapter 4

21 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 header QoS mechanisms: resource reservation (to make sure that Vo 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

22 Cisco Telephony deployment in a small company Chapter 5

23 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 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 phones and 1 legacy analog phone in the central office 1 Cisco phone and 1 analog phone in the fixed branch office 2 Cisco Aironet access points for a portable Wi-Fi Cisco telephone GateKeeper not necessary as all the phones registered to the same CallManager Voice ports Every 1760 router with 2 VIC modules and 2 voice ports per module (FXS and FXO)

24 LAB s components and logical topology Central Office (Poznan) Phone Location A Analog Phone PSTN Branch Office (Warsaw) Location B Analog Phone Phone Phone Phone Catalyst 3550 C1760 Access Router Wi-Fi AP Wi-Fi AP Cisco PIX NAT VPN tunnel C1760 Access Router VPN tunnel WAN Branch Office AP Roaming Cisco Call Manager Cisco App. Server Cisco VPN Concentrator Mobile Location C Wi-Fi mobile Phone Private networwork behind NAT Public Hotspot Cisco VPN Client Cisco Softphone Cisco Communicator Ms NetMeeting

25 VLAN configuration and physical inter-component connections PSTN Fa Fa Fa FXS FXO PBX FXO FXS Fa Fa Fa Fa Fa Fa Fa VLAN routing Serial (DTE) Serial (DCE) WAN / VPN Cisco Call Manager (MCS 7815) Application Server VLAN Internet VLAN Private (data) VLAN Voice VLAN CallManager VLAN Trunk

26 Dial plan architecture Central Office (Poznan) LOCATION A Branch Office MOBILE LOCATION C Branch Office (Warsaw) LOCATION B John Smith DN1: 1100 DN2: 1101 Kate Cole DN: 1102 Steve Edwards DN: 1103 Tom Jones DN1: 1200 DN2: 1201 Peter Hanks DN1: 1300 (NetMeeting) DN2: 1301 (NetMeeting) DN3: 1302 (Cisco Communicator) DN4: 1303 (Cisco Softphone) WAN Margaret York DN: 1140 FXS FXO FXO PSTN Kris Knight DN: 1220 FXS NAME: John Smith POSITION: Chairman PH. MODEL: C 7920 EXTENTION: 1100, 1101 EXTERN LINE: Central Office (Poznan) Kate Cole Steve Edwards Margaret York Secretary Technical Support Sales Manager C 7940 C 7902 Analog POTS Branch Office (Warsaw) Tom Jones Kris Knight Peter Hanks Chief of Staff Technical Support Sales Represent. C 7960 Analog POTS Ms NetMeeting 1200, Branch Mobile Office 1300, 1301, 1302, 1303 SERVICES Meet-Me-Conference 1016, Call Pickup 1015, Call Park 102X, Auto Attendant 1000, Integrated Contact Distribution 1005,

27 Simple voice connectivity scenarios John Smith DN1: 1100 DN2: 1101 Kate Cole DN: 1102 Steve Edwards DN: 1103 Tom Jones DN1: 1200 DN2: 1201 Peter Hanks DN1: 1300 (NetMeeting) DN2: 1301 (NetMeeting) DN3: 1302 (Cisco comm) DN4: 1303 (Cisco Softphone) WAN / VPN Margaret York DN: 1140 FXS FXO PSTN FXO Kris Knight DN: 1220 FXS Inter-office to - call (John Smith to Tom Jones) Inter-office Analog to - call (Kris Knight to Peter Hanks) Inter-office Analog to - Analog call (Kris Knight to Margaret York) to - PSTN call (Steve Edwards to TNC 2005 participient :)

28 PSTN backup Kate Cole DN: 1102 Tom Jones DN1: 1200 DN2: 1201 ISDN CCM WAN / VPN Margaret York DN: 1140 FXS FXO PSTN FXO Kris Knight DN: 1220 FXS Central Office Branch Office WAN is down or congested the phone at the remote office is losing 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,, backup is necessary to allow the remote phones coming back into operability

29 CT s supplementary services Chapter 6

30 Supplementary services overview Selected CT 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 CT s features and services

31 Software Conference Cisco CallManager John Smith DN1: 1100 Conference Controller Software MCU DN: 1016 CCM Peter Hanks DN3: 1302 (Cisco Communicator) Tom Jones DN1: 1200 WAN / VPN Meet-Me on Monday at 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

32 Call Pickup & Group Call Pickup Kate Cole DN: 1102 ROOM A Steve Edwards DN: 1103 ROOM B 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 phones: Call Pickup Group DN: 1015 CCM - 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. 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

33 Call Park Kate Cole DN: 1102 ROOM A Steve Edwards DN: 1103 ROOM B 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 CallManager cluster as well as between clusters. you can define either a single directory number or a range of directory numbers for use as call park extension numbers you can park only one call at each call park extension number Call Park DN range: Steve Edwards is answering a call from Kate s Cole 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 phone and is continuing the conversation

34 Cisco Phone services (1 of 2) CallManager Web and database server Web server HTTP/XML internet additional services let regard an 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

35 Cisco Phone services (2 of 2)

36 Cisco Telephony applications CallManager with Cisco Telephony directory CRA platform 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

37 Auto Attendant & Call Transfer CCM Auto Attendant Margaret York DN: 1140 Press 0 for the operator Kate Cole DN: 1102 Operator FXS Steve Edwards DN: FXO PSTN 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 Auto Attendant provides the following script: 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 Example: 2: PSTN-to Other to-company scenarios... 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

38 Cisco 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 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

39 Extension Mobility Kate Cole DN: 1102 CCM WAN / VPN 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 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 specific properties of a phone.

40 Other CT s features and services Cisco uone Voice Messaging The Cisco Unified Open Network Exchange (uone( uone) optional software, available as part of Cisco 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 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

41 Bibliography Chapter 7

42 Bibliography Margit Brandl, Dimitris Daskopoulos, Erik Dobbelsteijn, Jan Janak, Jiri Kuthan, Saverio Niccolini, Jorg Ott, Stefan Prelle, Sven Ubik, Egon Verharen, Telephony Cookbook TERENA Report, March 2004 Robert Padjen, Larry Keefer, Sean Thurston, Jeff Bankston, Michael E. Flannagan, Martin Walshaw, Cisco AVVID and 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, SYNGRESS Cisco CallManager Document - Release 3.3 Cisco Telephony Solution Reference Network Design Cisco CallManager Document - Release 4.0 Cisco Telephony Network Design Guide and websites

43 Questions?

44 Thank you

Enterprise VoIP. Silvano Gai. ftp://ftpeng.cisco.com/sgai/t2000voip.pdf. Cisco Systems, USA Politecnico di Torino, IT sgai@cisco.com.

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