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by Wojciech Nawrot, Wojciech Śronek, and Krzysztof Turza Poznań 2005

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

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by Wojciech Nawrot, Wojciech Śronek, and Krzysztof Turza Poznań 2005

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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. 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

  3. PBX replacement stages Chapter 1

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

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

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

  7. 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

  8. CIPT - Cisco IP Telephony Chapter 2

  9. 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

  10. A 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 (1 of 3) VoIP WAN Switch Gateway PSTN Router CallManager Cluster Cisco IP Phones IP Softphone

  11. A A • 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 (2 of 3) VoIP WAN Switch Gateway PSTN Router CallManager Cluster Analog Phone Cisco IP Phones IP Softphone

  12. A 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 Cisco IP Telephony components (3 of 3) VoIP WAN Switch Gateway PSTN Router CallManager Cluster Cisco IP Phones IP Softphone

  13. A A A CM Cluster CM Cluster CM Cluster Gatekeeper (CAC) 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 • a centralized Cisco CallManager solutionreduces equipment and operational expense and is a cost effective solution for for sites with less then 20 users ISDN PSTN PSTN IP WAN IP WAN backup Sec. voice path Sec. voice path Pri. voice path Pri. voice path • 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 Distributed Call Processing vs. Centralized Call Processing • Distributed Call Processing • Centralized Call Processing • dial backup is required for IP phone service across the WAN in case the IP WAN goes down

  14. 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

  15. Cisco IP phone physical connectivity and registration process • Physical connectivity: • 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) DHCP server Cisco CallManager + TFTP server 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

  16. VoIP signalling Chapter 3

  17. 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 H.225 (Q.931) H.245 T.120 RTCP RAS Control Data Audio Video Control Control G.7XX H.26X ISO Protocol LayerStandard Presentation G.711, G.729, G.729a, etc Session H.323, H.245, H.225, RTCP Transport RTP, UDP Network IP, RSVP, WFQ Link FR, ATM, ETH, etc RTP TCP UDP IP H.323 overview • H.323is an ITU-T recommendiation umbrella set of standards that defines components, protocols, and procedures necessary to provide audio, video, and data communications over IP-based networks

  18. H.323 Endpoints (Terminals) provide the user-to-network interfaces for H.323 protocol (IP phones or videoconferencing terminals) ISDN H.320 PSTN H.324 SIP • 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 H.323 Gatekeeper H.323 Gateway • 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 MCU • H.323 MCUs (Multipoint Control Units) provide conference support for three or more endpoints H.323 components

  19. IP phone IP phone PBX PBX Analog phone Analog phone FXOE&ME1/T1 FXOE&ME1/T1 SGCP for Cisco IP phones SGCPfor Cisco IP phones H.225, H.245, RTCP, RTP H.225, H.245, RTCP, RTP Direct dialing H.225, H.245, RTCP, RTP IP WAN CallManager /H.323 MCU H.323 Gateway CallManager /H.323 MCU H.323 Gateway RAS RAS RAS RAS Zone A Zone B RAS H.323 Gatekeeper(Zone A) 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) PSTN H.323 call stages and signalling flows

  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 3classification 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

  22. Cisco IP 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 • 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)

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

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

  26. Dial plan architecture Central Office (Poznan) LOCATION A Branch Office MOBILE LOCATION C Branch Office (Warsaw) LOCATION B John SmithDN1: 1100DN2: 1101 Kate ColeDN: 1102 Steve EdwardsDN: 1103 Tom JonesDN1: 1200DN2: 1201 Peter HanksDN1: 1300 (NetMeeting)DN2: 1301 (NetMeeting)DN3: 1302 (Cisco IP Communicator)DN4: 1303 (Cisco IP Softphone) IP IP IP IP IP WAN 6652921 IP IP 6652920 FXO FXO FXS FXS PSTN Kris KnightDN: 1220 Margaret YorkDN: 1140

  27. Simple voice connectivity scenarios John SmithDN1: 1100DN2: 1101 Kate ColeDN: 1102 Steve EdwardsDN: 1103 Tom JonesDN1: 1200DN2: 1201 Peter HanksDN1: 1300 (NetMeeting)DN2: 1301 (NetMeeting)DN3: 1302 (Cisco IP comm)DN4: 1303 (Cisco IP Softphone) IP IP IP IP IP WAN / VPN IP IP FXO FXO FXS FXS PSTN Kris KnightDN: 1220 Margaret YorkDN: 1140 • 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 :)

  28. IP ISDN PSTN backup Tom JonesDN1: 1200DN2: 1201 Kate ColeDN: 1102 IP CCM IP WAN / VPN IP IP FXO FXO FXS FXS PSTN Kris KnightDN: 1220 Margaret YorkDN: 1140 Central Office 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 analogphones are staying operational. • the PSTN is used as a backup path for voice connections • In the Centralized Call Processing scenario, IP backup is necessary to allow the remote IP phones coming back into operability

  29. CIPT’s supplementary services Chapter 6

  30. 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

  31. Conference Controller Conference Controller Software Conference Bridge @ Cisco CallManager CCM John SmithDN1: 1100 Peter HanksDN3: 1302 (Cisco IP Communicator) SoftwareMCU SoftwareMCU DN: 1016 Tom JonesDN1: 1200 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

  32. Call Pickup & Group Call Pickup • 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. Steve EdwardsDN: 1103 Kate ColeDN: 1102 ROOM B ROOM A IP 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

  33. Call Park • 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 Steve EdwardsDN: 1103 Kate ColeDN: 1102 ROOM B ROOM A IP IP IP CCM 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

  34. Web and database server CallManager Web server HTTP/XML internet http://Web_Server/Stockquote.asp?stock=TPSA Cisco IP Phone services (1 of 2) • additional services let regard an IP phone as a developed work tool • examples: • - personal address book • - corporate directory • - current stock value • - business information about client • web applications (ASP/JSP) returns XML objects to the phone

  35. Cisco IP Phone services (2 of 2)

  36. CRA platform CallManagerwith Cisco IP Telephony directory CRA Editor Softphone Cisco IP Telephony applications • Telephony Application Programming Interface (TAPI) • interoperability across various computer platforms – Java TAPI • CRA (Cutomer Response Applications) platform: • - CRA application server with CRA Engine • - CRA Editor and CRA administration web interface • - application scripts are stored in LDAP directory • - example of applications: auto attendant, integrated contact distribution

  37. Auto Attendant & Call Transfer • 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 CCM Auto Attendant Kate ColeDN: 1102 Steve EdwardsDN: 1103 Operator Press „0” for the operator IP IP IP 6652920 PSTN FXO FXS Margaret YorkDN: 1140 Example 2: Other scenarios ..... Example: PSTN-to-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

  38. Cisco IP Integrated Contact Distribution CCM+ ICD application Steve EdwardsTechnical SupportDN: 1103 • 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 • advantages (location independence, complete integration with CallManager, simplicity of installation, configuration and maintenance) Agent PSTN 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

  39. Kate ColeDN: 1102 Extension Mobility CCM IP IP WAN / VPN IP 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.

  40. 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

  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,„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

  43. Questions ?

  44. Thank you

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