VVT _05_2002_c2 2002, Cisco Systems, Inc. All rights reserved. 1

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3 Designing and Deploying Multi-Site IP Telephony Networks Session

4 Session Objectives Understand the underlying principles and components behind IP telephony Take a building block approach to the various IP telephony deployment models Empower the listener to be able to select, design and implement the appropriate model based on user requirements

5 Other Complementary Sessions VVT-214: Designing and Sizing IP Contact Center VVT-210: Designing and Deploying IP Telephony Applications NSC-215: Deploying QoS in an Enterprise Environment

6 Recommended Reading Deploying Cisco Voice over IP Solutions ISBN: Cisco IP Telephony ISBN: Available on-site at the Cisco Company Store

7 Recommended Reading Developing Cisco IP Phone Services: A Cisco AVVID Solution ISBN: Cisco CallManager Fundamentals: A Cisco AVVID Solution ISBN: Available on-site at the Cisco Company Store

8 Deployment Models

9 IP Telephony Design Goals Cisco CallManager Router/GW Regional Center Router Plus Voice Gateway Cisco CallManager PSTN Headquarters X IP WAN Router/GW Branch Office Rest of World Telecommuter

10 Single Site Deployments Single cable for phone and PC Inline power to phone sets Separate VLANs for voice and data Quality of Service CallManager clustering Up to 10,000 users per cluster Multiple clusters allowed via H.323 Gateways for PSTN access IP PSTN Apps (VM, UM, IVR, ) Call Manager Cluster

11 Independent Call Processing (No WAN Networking for Voice) A A Site A PSTN Site B All the characteristics of single site deployment Call processing at each site (independent) No limit to the number of sites Voice messaging and DSP resources at each site PSTN used for all external calls Compressed voice not required Uniform dial plan A Site C

12 Multi-Site Deployments Distributed Call Processing PSTN IP WAN Branch B Headquarters CallManager and voice messaging at each site networked together Over 100+ sites Gatekeeper-based admission control WAN QoS Compressed voice over IP between sites Transparent use of PSTN if IP WAN unavailable 5293_05_2002_c2 2002, Cisco Systems, Inc. All rights reserved. Branch C 12

13 Multi-Site Deployments Centralized Call Processing Branch B PSTN Branch C Headquarters IP WAN CallManager and applications at central site Centralized dial plan and administration Up to 10,000 users total per cluster CM locations-based Call Admission Control Survivable Remote Site Telephony (SRST) for branch offices

14 Building Blocks and Components

15 Building Blocks Single site call processing Highly Available data network Applications IP phone connectivity Gateways Call processing Security Resources Independent call processing Distributed call processing Centralized call processing

16 Single Site Deployments Overview CallManager, applications and DSP resources at same physical location Supports up to 10,000 IP Phones per cluster Multiple clusters can be interconnected LAN QoS PSTN used for all external calls Applications (VMail, IVR, ICD,...) CallManager Cluster PSTN

17 How It Has to Be Built Applications Telephony Infrastructure Network Infrastructure Standards-based Personalized Scalable Flexible QoS-enabled Highly available

18 Campus Infrastructure Highest Level of Redundancy Access Layer Per-VLAN Spanning-tree Access Layer 2 UplinkFast Distribution Layer HSRP w/ load balancing Distribution Layer 3 Core Layer 3 Server Farm OSPF/EIGRP configured for fast convergence Core OSPF/EIGRP configured for fast convergence Distribution Layer 3 Access Layer 2 WAN Internet PSTN

19 Quality of Service (QoS) Features PSTN IP WAN Campus Branch Office Campus Access Inline Power Multiple Qs 802.1p/Q CoS DiffServ ToS Extended Trust Campus Distribution Multiple Qs 802.1p/Q CoS DiffServ ToS Policing Congestion Avoidance (WRED)

20 3 Ways to Power IP Phones Inline power Powered linecards for Catalyst switches and 2600/3600/3700 routers Uses pairs 1 and 2 (same as Ethernet) for delivering 48V External power Needs external power patch panel Patch panel delivers 48V over pairs 3 and 4 Wall power Needs DC converter for connecting IP phone to wall outlet Combination of Ways Can Be Used for Redundancy

21 Catalyst Switch <-> Phone Interaction Plug-and-Play Operation 1. Phone Discovery 2. Provide Power 3. CDP Unpowered phone plugs into powered line-card port Port senses the device using phone discovery mechanism and reports it to the supervisor Supervisor checks power budget, allocates default amount, and informs port to apply power Port turns on power to the phone and reports linkup to supervisor, after the PHY on the phone is enabled If phone was powered by external patch panel or wall power, switch port reports linkup to supervisor Phone begins CDP exchange with the switch and gets its VLAN ID (VVID); phone reports actual power needed for operation Phone now sends a DHCP request on that VLAN for an IP address

22 Catalyst Auxiliary VLAN This Feature Provides Automatic Phone VLAN Configuration Tagged 802.1p/q Phone VLAN = 200 PC VLAN = 3 IP Subnet B Untagged IP Subnet A No end-user intervention required Provides the benefits of VLAN technology for the phone Preserves existing IP address structure Uses 802.1P/Q technology between switch and phone

23 Phone Actions on Startup 1 2 Use Any One 3 4 Get IP address, mask, DNS, etc.* Static or DHCP Get TFTP server address* Static address Option 150 (single IP address) Option 66 (first IP address or DNS name) Look up CiscoCM1.your.domain Get configuration from Cisco CallManager TFTP* List of up to three Cisco CallManagers Region info and keyboard template Version of code to run Get new code (one time only) 5 Register with Cisco CallManager DHCP DNS *Uses last known good working profile if servers fail to respond 1 2 Cisco CallManager TFTP 3 4 5

24 IP Addressing Deployment Options IP Phone + PC on Same Switch Ports IP Phone + PC on Same Switch Ports IP Phone + PC Share the Same Device (Soft Phone) Real IP Addresses IP Phone + PC on Separate Switch Ports IP Phone Uses Network Real IP Addresses IP Phone + PC on Separate Switch Ports Real IP Addresses IP Phone Uses Network

25 Basic Phone-to-Phone Call Processing 1 Call Setup 4 Ring Back 2 Cisco CallManager E.164 Lookup IP WAN 6 Connect RTP Stream Call Setup 3 4 Ring VSM PSTN 5 Off Hook Media stream is released to the endpoints Call preservation is maintained if CM fails* *Only for SCCP, MGCP and TAPI/JTAPI. Not for H.323

26 CallManager Primary Functions Signaling and Device Control (IP Phones, Network Gateways, etc). Operation, Administration, Maintenance and Provisioning (OAM&P) Functions (PBX and more ) Call setup, teardown, supervision IP phone registration, Device load management, MGCP GW Call Agent Protocol translation Dial plan, call routing Call Detail Recording (CDR) CTI integration (TAPI/JTAPI) Integrated Services: Media Termination Point (MTP) Music on Hold, IPMA, Ext. Mobility Conference bridge SW or HW Telephony Call Dispatcher TFTP/DNS/DHCP server Integrated HTTP/XML server CTI Manager (TAPI/JTAPI) Protocols Supported: Cisco SCCP H.323v2 MGCP PBX/PSTN protocols (CAS, CSS ) TAPI/JTAPI Future: H.323v3, SIP, QSIG Communicates with: IP phones/soft phones Media Servers (MoH, Conf, Xcoders) Other CallManagers Legacy PBXs Gateways, Gatekeepers Voic /UM servers CTI applications

27 Media Convergence Servers High Low Availability Performance Pentium III GHz 1 GB error correcting RAM Up to 1,000 phones Components Single 40 GB Fast ATA hard drive Single power supply Platform 1U rackmount chassis Desktop stack option to stack up to 6 servers MCS Performance MCS Performance Pentium III 1.26 GHz 1 GB error correcting SDRAM Up to 2,500 phones High Availability Components Dual 18.2 GB Ultra3 hot swap SCSI Hard Drives Redundant hot swap power supplies Hardware RAID controller (RAID 0/1 disk mirroring) Optional 20/40 GB hot plug DAT tape drive Platform 2RU rackmount chassis

28 How Many CallManagers Do I need? Depends on: Model of server Number of IP phones Number of gateways Number/Type of Applications Level of redundancy required Cisco IP Phones Applications CallManager PC Stations Catalyst Switch Router/GW IP WAN PSTN

29 Call Manager Redundancy Group Primary Secondary Last Resort Every device has a prioritized list of up to three CallManagers to which it can connect (except H.323) Part of TFTP download Determined by Device Pool Active TCP connection to primary and secondary servers Automatic fail over and fall back

30 Call Processing Scalability/Redundancy Clustering Options Primary Secondary/ Backup Tertiary/ Last Resort Up to 10,000 IP Phones per cluster* Other optional servers not shown (redundant TFTP, MoH, etc.) To 2,500 IP Phones To 5,000 IP Phones To 10,000 IP Phones Publisher and TFTP Server Publisher and TFTP Server Publisher, Backup and TFTP Server Primary CM 1 to 2500 Backup 1 to Backup Backup 1 to to to to Greater than 10,000 in next release

31 Shared Media Resource Types SM Viper MTP Call Manager Transcoding Call Manager MTP H.323v1 G711,G723 Music on Hold x1000 Conferencing Call Manager MOH Server Call Manager x2000 Media resources register with up to three CMs Media resources are shared/ advertised across all CMs in a cluster (any phone can use any resourc on any server at any time) Topological awareness/ resource pooling via Media Resource Groups/ Lists

32 opologically Aware Media Resources xample MRG/MRGL Configuration MOH1 Dallas MTP1 SW-CONF1 Router HW-CONF1 XCODE1 IP WAN San Jose CM2 MTP2 MOH2 SW-CONF2 Router XCODE2 HW-CONF2 DallasSoftware MTP1 MOH1 SW- CONF1 DallasHardware XCODE1 HW- CONF1 SanJoseSoftware MTP2 MOH2 SW- CONF2 SanJoseHardware XCODE2 HW- CONF2 Created 4 MRGs and assigned all resources to groups as shown. Created a DALLAS MRGL with the following groups DallasSoftware:DallasHardware:SanJoseSoftware:SanJoseHardware Created a SanJose MRGL with the following groups SanJoseSoftware:SanJoseHardware:DallasSoftware:DallasHardware Assigned phones in Dallas to use DALLAS MRGL and phones in SJ to use SanJose MRGL With this arrangement, local media resources will always be used first, and will fail over to remote resources only if local resources are unavailable

33 DSP Resource Considerations Resources are required for conferencing, transcoding, and MTP Strongly recommended these are provisioned using hardware resources. Software a potential security hole and G.711 only Conference typically 5% of user count Transcoding based on number of WAN connections possible DSP resources should be over-provisioned Shared resources provides Loadsharing and Redundancy Backup Backup Publisher and TFTP Server Conf Conf Xcode Xcode

34 Transcoding Between Clusters With CCM 3.1 and above, transcoding resources are used only where required by an end point Transcoding Voice Mail CallManager Cluster Cisco IOS Gatekeeper CallManager Cluster Voice Mail Router/GW IP WAN Router/GW

35 Additional Design Considerations for Media Resources DSP resources can be shared and distributed using MRGs/MRGLs only in CM 3.1 and above With CM 3.1 and above, HW and SW based conferencing resources can now be provisioned AND used at the same time Conferences and MTP/Transcoding sessions can NOT span across multiple DSPs Know the conference call, transcoding and MoH usage patterns for your network Some applications are G.711 only (Personal Assistant, IVR, ICD, Conference Connection)

36 Cisco CallManager Music-on-Hold Service Music-on-Hold application installed to Media Convergence Server Scalable, redundant with multiple Media Convergence Servers Audio sources: 50 continuously looping files One fixed source (optional) MC or unicast per source 250 simultaneous streams per server (25 with collocated CallManager) G.711, G.729a, and wideband audio per source File-1 File-2 File-3 File-4 src-1 src-2 src-3 IP Phones File-5 src-4 File-6 src-49 File-99 src-50 Sound Card src-51 SoftPhones Gateways

37 Cisco CallManager Extension Mobility Service XML-based login application for 7960/7940 phones Authenticates username/pin via LDAP directory Updates CallManager publisher database when user logs in User s profile is pushed to phone where login occurred Phone takes on user s line appearances and personal settings Applicable for mobile office environments and roaming Integrated in CallManager 3.1 and above EM service can be installed on a separate MCS or collocated with CallManager Directory Web Server Bldg 2 Bldg 1 Call Mgr Data Center DHCP Server

38 Cisco CallManager Corporate LDAP Directory Integration LDAP Corporate Directory Access (User Search) Integration (Profile Storage) eb erver Endpoints (IP phones, SoftPhone) CallManager Cluster(s) Applications (IVR, ICD, PA) Voic / UM (Unity) CallManager/Applications use LDAP to store username/pin and user/device profiles Endpoints/Applications access LDAP to search for users and authenticate logins Supported directories:

39 Choosing The Right Gateway Catalyst 4000 Catalyst 6000 Cisco 7200-VXR ICS 7750 Cisco 2600 Cisco MC3810 Cisco 3600/3700 Cisco AS5300 Cisco AS5800 Cisco 1700 Cisco VG200 Cisco VG248

40 Choosing The Right Gateway Roles, Platforms and Protocols PSTN CallManager Cluster IP WAN Remote Site Central Site Gateway Role: Standalone or integrated Router/GW/SRST Protocol MGCP*, H.323 Target Platforms: VG200 (standalone) Catalyst xx/36xx/37xx MGCP, H.323 Platforms: WS-X6608, WS-X6624, Cat AGM VG xx, 36xx/37xx 7200, 7500, AS5300 MGCP and SCCP gateways can be centrally configured from CallManager

41 Choosing The Right Gateway MGCP vs. H.323 Gateways CM-E X CM-D PSTN CM-A CM-C Gateway CM-B MGCP GWs rehome to secondary/ tertiary CallManager H.323 GWs fall back to alternate VoIP dial-peer MGCP call-agent ccm-manager redundant-host [no] call-manager redundancy switchback [immediate graceful delay delay-time] H.323 dial-peer voice 101 voip destination-pattern 1111 session target ipv4: preference 0 voice class h323 1 dial-peer voice 102 voip destination-pattern 1111 session target ipv4: preference 1 voice class h323 1 voice class h323 1 h225 timeout tcp establish 3

42 Choosing The Right Gateway MGCP PRI Backhaul An internal interface between CCM and Cisco gateways (i.e. separate channel for backhauling signaling information) Layer-3 PRI (Q.931) is backhauled over TCP Independent of the native protocol over PSTN TDM interface Uses Q.931 PRI protocol for call control and MGCP protocol for media control Separate channel for MGCP and PRI backhaul MGCP over UDP, PRI backhaul over TCP Transparent to customers and end users MGCP UDP Cisco CallManager Gateway Q.931 Backhaul TCP PBX With CM 3.1 and above, GW terminates Layer-2 signaling; If CM goes down, signaling Is maintained at the GW and It just starts using its secondary CM

43 IP Telephony Security Considerations CallManager Minimize Win2K services NTFS Secure IIS Lock down SQL IDS/virus A Endpoints Use separate addressing for voice and data RFC1918 is preferred CCM Firewall Allow only call control, LDAP, management Control source addresses Outside World No NAT across Internet IOS DoS tools Use sensors Internet IP WAN PSTN Campus Network Secure access (TACACS+, SSH, Radius) Use VLANs Use IP filters between voice and data network

44 Host-Based IDS Certification Cisco Intrusion Detection System (IDS) Host Sensor Purchased separately from Cisco* CallManager, operating system, web server security Senses and reacts to attack on application, OS, IIS One component in the voice security solution Certified for CallManager 3.0, 3.1 and 3.2 *Will be bundled in a future release

45 Virus Checker Certification McAfee NetShield version 4.0 Certified on CallManager versions 3.0, 3.1 and 3.2 Very specific caveats regarding scanning effects on CallManager performance (1) schedule scans during off-hour periods (2) disable heuristic scanning

46 Building Blocks Single Site Call Processing Independent Call Processing Distributed Call Processing Centralized Call Processing

47 Independent Call Processing (No WAN Networking for Voice) A Site B A Site A PSTN All the characteristics of single site deployment Call processing at each site (independent) No limit to the number of sites Voice messaging and DSP resources at each site PSTN used for all external calls Compressed voice not required Uniform dial plan Site C A

48 Building Blocks Single Site Call Processing Independent Call Processing Distributed Call Processing WAN QoS Inter-Cluster Trunks Gatekeeper-Based Call Admission Control Distributed resources Centralized Call Processing

49 Distributed Call Processing Deployments Overview Applications (VMail, IVR, ICD,...) CallManager Cluster CallManager Cluster PSTN Applications Headquarters GK Gatekeeper IP WAN CallManager Cluster Branch A CallManager and applications clustered at each site (depending on application) Up to 10,000 IP phones per cluster 100+ sites Transparent use of PSTN if IP WAN unavailable Branch B Application

50 Quality of Service (QoS) Features PSTN IP WAN Campus Branch Office Campus Access Campus Distribution WAN Aggregation Branch Router Branch Switch Inline power Multiple Qs 802.1p/Q CoS DiffServ ToS Extended Trust Multiple Qs 802.1p/Q CoS DiffServ ToS Policing Congestion avoidance (WRED) 802.1p/Q CoS DiffServ ToS LLQ Traffic shaping Policing Link efficiency (LFI, crtp) WRED 802.1p/Q CoS DiffServ ToS LLQ Traffic shaping Policing Link efficiency (LFI, crtp) WRED Inline power Multiple Qs 802.1p/Q CoS DiffServ ToS Extended Trust

51 Inter-Cluster Call Processing Beyond 10,000 IP Phones: Inter-Cluster Trunks Inter-Cluster calls use H.323 H.323 Inter-Cluster Trunks defined to all other clusters (full mesh) Maximum of 1000 H.323 calls on each Inter-Cluster Trunk (500 in 3.1 and below) H.323 Inter-Cluster Trunks Cluster 1 Cluster 2 Cluster 3 Cluster 4

52 Inter-Cluster Call Processing Multiple Inter-Cluster Trunk Gateways Cluster 1 Cluster 2 Backup 1 Route Groups 1 Backup Publisher 2 2 Publisher Backup 3 3 Backup 525-xxxx xxxx Inter-Cluster Trunk GW for each call processing server Maximum of 6,000 inter-cluster calls (1000 calls per trunk * 6 servers/trunks per cluster) Slow fail-over within the Route Group due to TCP connect timeout Cumbersome configuration Practical limit: 10 clusters Cluster 3 Backup Backup Publisher 527-xxxx

53 Inter-Cluster Call Processing Inter-Cluster Trunks Using Anonymous Device Gatekeeper Cluster 1 Backup 1 GK 1 Cluster 2 Backup Publisher Backup 2 3 Gatekeeper/ Anonymous Device 2 3 Backup Publisher 4 4 CM Registers with Gatekeeper and Anonymous Trunk defined Calls routed to currently registered CM Fast fail-over if a server is unavailable (Gatekeeper tracks registration status) Maximum of 1000 inter-cluster calls (only one Gatekeeper/Anonymous Trunk per cluster in 3.2 and below)

54 Need for Call Admission Control Protecting Voice from Voice Example: WAN Bandwidth Can Support only 2 Calls What Happens when Third Call Attempted? Cisco CallManager Call #1 Call #2 Call #3 IP WAN X X X Cisco CallManager Call #3 Causes Poor Quality for ALL Calls Need to Prevent Third Call from Traversing IP WAN

55 Call Admission Control 100+ CallManager Clusters Hub-and-spoke topology 1 CM cluster per Zone Maximum of 100 zones per GK Approximately 50 Gatekeeper calls/sec. per GK CAC HSRP for GK redundancy Locations CAC used for remote sites with no Locations CAC CallManager (centralized model)... Zone 1 Gatekeepers GK GK Zone 2 Zone 100

56 Call Admission Control Beyond 100 CallManager Clusters Each GK controls up to 100 Zones A Directory GK controls all GKs and contains dial plan information Each Directory GK controls up to 2,000 zones More hierarchy levels can be added above the firsttier Directory GK Only limitation is hub-and-spoke topology Directory Gatekeeper GK Zones GK... GK Zones 1,901-2, Zone Zone 2 Zone 2,000

57 Building Blocks Single Site Call Processing Independent Call Processing Distributed Call Processing Centralized Call Processing Transcoding and Conferencing Locations-Based Call Admission Control Survivable Remote Site Telephony and Gateways Clustering over the WAN Dial-Plan Internationalization

58 Centralized Call Processing Overview Applications (VMail, IVR, ICD,...) CallManager Cluster PSTN SRST-Enabled Router IP WAN Branch A Headquarters SRST-Enabled Router CallManager at central site Applications and DSP Resources can be Branch B centralized or distributed Supports up to 10,000 IP Phones per cluster Locations-based Call Admission Control Survivable Remote Site Telephony for remote branches

59 Centralized Call Processing Benefits Simplified, centralized management and administration Lower initial investment and maintenance costs Seamless WAN connectivity of all remote branches (toll bypass savings) Unified dial plan Basic call processing available at remote branches even in case of IP WAN failure

60 Conferencing and Transcoding Resources VSM Compressed Call Leg G.711 Call Leg Voice Services Module Cisco Unity VM/UM Server Cisco CallManager Cluster VSM Router/GW Conferencing Cisco Unity VM/UM Server Cisco CallManager Cluster Transcoding Router/GW IP WAN VSM

61 Distributed Conferencing Resources MRL 1. Br1 2. HQ1 3. HQ2 X PSTN CallManager cluster MRL 1. HQ1 2. HQ2 IP WAN Device Pool Branch Conf MRG=Br1 MRG = Media Resource Group MRL = Media Resource List Conf Conf MRG=HQ1 Conf Conf MRG=HQ2 Device Pool HQ Conference between A, B and X no voice across WAN Requires Conferencing/Transcoding Resources at branch (i.e. VG200 or NMHDV DSP Farm module) and CM 3.1(2) or above No conferencing during WAN failures

62 Call Admission Control Locations Construct Prevent WAN link oversubscription by limiting voice bandwidth Assign bandwidth limit for voice per location (in kbps) When resources are insufficient, phone gets fast-busy tone and a message is displayed Automatic Alternate Routing (AAR) in pending release PSTN 2 1 STOP Location 1 IP WAN Central Site Remote Sites Location 2 Location 3 Max BW = 80 kbps Max BW = 160 kbps

63 Location-Based Admission Control and Cluster Interaction One CallManager OR Two CMs in a Cluster where All Phones Register to Same CallManager Central Site CM-A Central Site A Primary CM Backup CM B Locations WAN Cluster CM-E CM-D CM-C CM-B Remote Sites Location 1 Location 2 CM 3.0 Maximum 2,500 Phones per Cluster Remote Sites Location 1 Location 2 CM 3.1+ Maximum 10,000 Phones per Cluster

64 Survivable Remote Site Telephony Centralized CallManager Cluster SRST Site 2 Location 2 = 128kbps Max Router/ GW PSTN IP WAN Router Location 1 IP WAN X SRST Site 3 Location 3 = 256kbps Max Requires a Cisco CallManager for Normal Operation SRS Telephony in Cisco IOS Router for Fallback Call Processing Supported on 17XX, 26XX, 36XX, 37XX, 72XX and Catalyst 4224 IP WAN Router call-manager-fallback ip source-address port 2000 max-ephones 5 max-dn 10 keepalive 60

65 Survivable Remote Site Telephony How It Works Normal WAN Operation Failure Signaling Traffic SRST router ISDN Backup Signaling Traffic Data Traffic CallManager cluster IP WAN Remote Site Voice Traffic PSTN Voice Traffic Central Site Phones reregister with their default router Subset of features available (DID, DOD, Hold, Transfer, Speed Dial, Caller ID) ISDN backup may be used for data traffic only ACL needed on branch router to block SCCP traffic

66 SRST Phase I Features Available in 12.1(5)YD System Features Rehoming of IP phones upon failure to branch router for call processing Maintain local extension-toextension calls upon failure Maintain extension-to-pstn calls upon failure Maintain existing calls upon recovery Support for IP and POTs phones DID and DOD calling Caller ID and ANI support Calling party name Call detail recording/radius server Interworking with Cisco Gatekeeper Voice mail support through local answering machine Phone Features Up to six lines per phone Up to 48 line appearances per system (144 if a 3660 router is used) Primary line on phone Speed and last number dial Distinctive ringing Transfer (without consult) Call hold/call retrieve Call waiting Trunk Features PSTN T1 and E1 CAS trunks support Analog FXS and FXO BRI (euro ISDN) WAN link support: Frame Relay, ATM, MLPPP, Serial, AAL2, and DSL

67 SRST Phase II Features Now Available in 12.2(8)T Music/Tone on hold and on Transfer (MOH only for PSTN endpoints) Distinctive ringing Internal vs. external ISDN Primary Rate Interface support Alias Lists Route calls meant for central site target to a selected local destination Translation Rules Allows user speed dials to be expanded to use PSTN during WAN outage Transfer to centrally located voic system Class of Restriction Prevent lobby phones from making expensive calls during WAN outage H.323-based transfer across Cisco IOS endpoints

68 SRST Supported Platforms Max IP Phones Per Platform Max # of IP Phones Supported Catalyst 4224 Cisco 3640 / 3724/ /IAD2400/MC3810/2600/3620 Cisco 3660 Cisco

69 emote Site Gateway ormal MGCP Operation Branch Office Translate FXS/FXO/T1CAS signaling to MGCP msg, and backhaul ISDN L3 signaling to CM POTS FXS FXO PSTN Main Site Call Manager Call Processing and Routing Control T1-CAS T1/E1 PRI MGCP GW WAN MGCP Media and Resource Control In MGCP mode, the gateway translates FXS, FXO and T1-CAS signaling into MGCP messages that are sent to the CM for call control ISDN PRI Layer 3 signaling is back-hauled through the gateway directly to the CM for processing The main site Call Manager handles call processing and routing, and uses MGCP to control media on the remote gateway s telephony interfaces

70 emote Site Gateway GCP Gateway Fallback* Branch Office Gateway Fallback MGCP H.323 POTS FXS FXO PSTN Main Site Call Manager MGCP GW T1-CAS T1/E1 PRI XWAN When the MGCP Gateway loses contact with all of its CallManagers, it will fallback to H.323 control to support basic call handling of FXS, FXO, T1-CAS and T1/E1 PRI interfaces, ie no supplementary services *SRST and MGCP Fallback cannot run on the same router until 12.2(11)T

71 Call Processing Clustering over the WAN Publisher/ TFTP < 40ms RTD QoS Enabled BW 40ms RTD between ANY two CallManagers 900 kbps for each 10,000 BHCA within the Cluster Four Active Locations, Maximum (4 active CM s) Failover across the WAN supported (Additional BW) Check out the IP Telephony Design Guide for CallManager 3.1 and 3.2 for Full Details

72 Call Processing Spatial Redundancy CallManager cluster Voice Mail Server Voice Mail Server IP Phones L.A. Space IP Phones San Diego Spatial Redundancy = Resilience Single Point of Administration, Extension Mobility, Feature Transparency and Unified Dial Plan Can Be Used for Business Continuance

73 Call Admission Control Locations for Centralized Call Processing San Jose 2 Data Center allmanager Cluster <None> Gatekeeper GK SJ HQ 2 Mbps Need to preserve hub-and-spoke topology Leave devices in San Jose Data Center in the <None> Location, assign all other sites to Locations (up to 500) Remember: no automatic PSTN fallback in 3.2 and below If BW not available, must hang up and dial PSTN access code = Loc. A Loc. C Seattle 72 kbps L.A. 240 kbps Phoenix 120 kbps Location <None> Loc. D

74 Dial Plan Goal Transparent Automatic Route Selection User Dials Does the WAN Have Sufficient Resources to Place Call? Yes Call Manager CM Strips 1st 3 Digits for IP WAN X 1212 Sent to Remote Site Proper QoS Ensured IP WAN IP Phones User Dials Reference: VVT-410 Dial Plan Session Router/GW CM Inserts 1408 for PSTN Dynamic Alternate Routing PSTN Sent to PSTN IP WAN Goes down or Not Enough Resources

75 The Dial Plan is the IP Routing of IP Telephony Networks Router IP Route 1 st Choice 10.1.X.X 10.2.X.X IP Route 2 nd Choice Local Hosts Routing Table 1000 Call Manager XXXX XXXX IP WAN 1 st Choice 1001 Router/GW PSTN 2 nd Choice

76 Call Manager Dial Plan Architecture Route Patterns Match an E.164 address range or specific address. Route Pattern Digit Manipulation, Route Filtering Route Lists How to reach a destination via prioritized route groups. Route Groups Form a prioritized Trunk Group by pointing at devices. Devices Assigned in Route Groups 1. Gateways 2. Remote Call Managers 1st Choice Route Group IP WAN Route List No No Try 2nd Try 3rd Choice Choice (If Available) Route Groups Point to Devices 1st Choice A 2nd Choice Route Group PSTN 2nd Choice

77 Call Restrictions Creation of Dial Plan Policy Groups Partition. Devices with similar reachability characteristics. Subset of all callable numbers. Items placed in Partitions: Directory Numbers (DNs), Route Patterns, CTI Route Points Dept A Dept B Calling Search Space. Set of Rules Which Partitions a device may search in for a dialed number. Provides dialing permissions/restrictions. Each device assigned a Calling Search Space Dept C Dept D Internet PSTN Call Manager

78 Example Use of Partitions and Calling Search Spaces (Calling Restrictions) San Jose A Partition Assignment SJ-Users = All SJ IP Phones SJ-PSTN = 9 Route Pattern PSTN Calling Search Space Unrestricted = SJ-Users, SJ-PSTN SJ-Only = SJ-Users Employee Phones Lobby Phones IP Phone Calling Search Space Assignment Staff IP Phones = Unrestricted Lobby IP Phones = SJ-Only Employees May Dial Anywhere Lobby Phones Can Dial Internal to SJ

79 Summary

80 Summary Plan and stay within design guidelines (inline power, IP addressing, dial plan, voice mail, combined telecom/data teams, etc.) Converged networks require an intelligent QoS-enabled infrastructure End-to-end IP telephony solutions are here now Week long CIPT training available from certified partners Visit the World of Solutions and the Design Clinic to learn what Cisco products can be used to build your IP Telephony network today

81 Designing and Deploying Multi-Site IP Telephony Networks Session

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