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Converged Network Design

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Agenda Consolidation Cisco IP Telephony IP Telephony Components But First . Quality Of Service (QoS) IP Phone Roadmap Call Manager 3.0 Distributed Call Processing Catalyst Enhancements So how does this all work then Cisco AVVID Application Integration Design Guide Summary

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Consolidation PCs Distribution Server Farm Wiring Closet VoiceMission- Critical Application Video Surveillance Mission- Critical Application Web Servers PCs, Voice, Video, Business Apps Four Different Traffic Types - all Mission Critical PCs, Voice, Video, Business Apps

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Key Technology Requirements for IP Telephony Reliability Dial tone is always there Quality of Service WAN and LAN, good quality voice always! Power & Current Infrastructure Integration eg: cabling, integration with current PBX Scalability to Large Campus Sizes Need to move beyond the current 200 user limit Application Integration - Current and Future Call processing, Unified Messaging, Call center etc. integration in the Enterprise space is fundamental

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Cisco AVVID An End-to-End Architecture Softphone MessageServer Telephony Application Servers PagingServer DirectoryServers ContentServer ContentServers IP Phones PCs Video Distributed Adaptive Distributed Adaptive SERVICE SUPPORT GatewaySwitch Router Cisco IP Fabric Intelligent Network Services Platforms Infrastructure ClientsApplications Open Manageable Open Manageable MessageServers CallManagerServers

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PSTN What do I need for IP Telephony? Phones Supporting, H.323, SIP or a low weight stimulus response protocol such as Skinny e.g. Symbol, Cisco Gateways PBX and PSTN connectivity Applications and Call Processing CallManager, voice mail, IVR, etc. Network infrastructure Routers, switches, wire, WAN services

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Where do all these bits fit? WAN PBX PSTN Cisco IP Phones: 12SP+/30VIP Next Generation Application Integration: Voice Mail Dialing Plans Supplementary Services Cisco CallManager LAN Switches provide QoS, inline Power and Switched Infrastructure Gateways Provide Analog or Digital Access and QoS support

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Open, Standards-Based Gateway Cisco Directory Server CallManager Legacy PBX Voice-Enabled Router Cisco IP Phones QoS Enabled Catalyst Switch Application Servers IP PSTN

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Open System Applications New World Ecosystem IP Gateway CallManager QoS Enabled Catalyst Switch Application Servers Telekol Picazo Meet Me Conference PSTN Voice Mail/UMS from Amteva Acquisition Call Center Applications from GeoTel Acquisition

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Cisco IP Phones Firstly they look like phones... Soft key & display based access to features and value-added services Programmable soft buttons No paper labels, easy set installation / relocation Integral 10BaseT Hub Locally powered Range of Voice services G.711 & G.723.1

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and if you have a lot of legacy phones Support via H.323 & Skinny Connect existing circuit switched phones to an IP network. Supported phones Lucent Definity all 8400, 6400 series and Call Master 3 Nortel Norstar 7324, 7310, 7208 and 7100, Meridian M2006, M2008, M2009, M2012, M2216, and M2616 Mitel SX-50, SX-200,SX-2000 Superset 410, 420, 430 (DNIC), and 4000 series Siemens Hicom 110E,150E, 300E All Optiset E, Rolm 9751 RP200, RP300, RP400, and RP600 Ericsson MD110, Business Phone Dialog 3000 Still waiting for BU decision on product

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Cisco PSTN Gateways All Cisco IOS/CatOS Platforms The link from IP to Circuit Switched - H323 Gateway Both Digital & Analogue support Support for Supplementary Services (Hold, Transfer, Conference, etc) Available today. PBX PBX PSTNPSTN PBX

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5300 7202 72047206 7505 75077513 Small Branch Enterprise Branch Central Site/ Regional Branch Headquarters 1750 Cisco Multiservice Gateways Performance Scalabiliy and Density2600 3620 36403660 3810 DE 30+

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Cisco Call Manager Provides intelligent Call Processing & PBX functionality Advanced Call Control Scalable and on open systems Fault tolerant Browser Accessed Standards based H323 and MGCP support Application Integration Unified Messaging Call Centres

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Quality of Service - What is it? QoS = Preferential treatment QoS prioritizes traffic into service levels and provides preferential treatment to some traffic at the expense of lower priority traffic Needed in both the LAN and the WAN All traffic gets the same service We now have the possibility of three types of traffic, all delay sensitive A traditional network is best-effort ! This will not work with a heavily loaded Voice enabled network

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.1 G.711 G.729(a) G.723.1 64k + Header 8k + Header 6.3k/5.3kk + Header 79.5k (on Ethernet) 20k/10k (on PPP) 18k/8k (on PPP) So How Much Bandwidth Does A Voice Call Really Need ? Voice calls without Compressed Real Time Protocol can take up as much as 80K on a WAN link when using G.711 Administrator can defines which CODEC in regions Currently no low-bitrate conferencing

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Challenge Cisco Solutions Packet Residency Slow Link Freeze-out by Large Packets Interleaving FRF.12, MLPPP, IP MTU Size Reduction, Faster Link Bandwidth Consumption Header Size on Low Bandwidth Links Compression Codecs, RTP Header Compression, Voice Activity Detection WAN Oversubscription, Bursting Traffic Management Router Traffic Shaping to CIR, High Priority PVC, Data Discard Eligibility Congestion Delay and Delay Jitter Intelligent Queuing WFQ, IP Precedence, RSVP, Priority Queuing VoIP Low Speed Link (

Cisco CallManager Cisco uOne Messaging Server IMAP4 Message Store 5201 5202 CFNA-->5050 1-Call Setup 2-E.164 lookup 3-Call Setup 4-Alerting (Ring) 4-Alerting (Ringback) 5-No Answer 8-Media Connect 5050 6-Call Setup 7-Auto-Offhook 11-Greeting/tone playback (SMTP) 13-Message Waiting 14-Message Waiting 11-Greeting/tone playback (SMTP) 12-Record Message (RTP) 12-Record Message (SMTP) LDAP v3 Directory 9-Get Subscriber 10-Set Subscriber All signaling and media over IP Cisco CallManager Voice Mail call control 5201 dials 5202 - fwd to VM

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Application Integration When my wife calls to tell me she's going into labour, it's no big deal for an IP-based switch to look into my scheduler, find the conference room I'm in and forward the call there. Getting a traditional PBX to do that would take a whole fleet of Andersen consultants. Christian Renaud, Cisco EVBU Product Marketing 27th September 1999

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Ring uOne Check My Calendar Cell / PDA IP Network If Its a Customer, Ring Me Wherever! Ring If Sister Calls and I Am in a Meeting, Send to Unified Messaging If Its the Boss, and Im not in a Meeting, Ring Me but Dont Call Cell Ring Virtual Intelligent Assistant

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Design Caveats Maximum 200 users per CM No WAN connectivity Call Manager cannot use Gatekeeper for Address Resolution (Does not enhance dial plan scalability) No two sites may have same internal dial plan Therefore only 1 instance of 1XXX, 2XXX, 3XXX etc Limited to 10 Site Deployments Call Manager uses 128kbps in the ARQ (Only going to use 80kbps for G.711) Call Manager registers EACH GK Controlled H.323 device as a separate VoIP-GW with Gatekeeper

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Router/GW PSTN V Site A Site B Site C IP WAN Router IP WAN Router Call Manager V V Call Manager Call Manager Call Manager at each site 200 IP Phones 100 Voice Mail Boxes GW Selection based on PSTN signaling PRI - DT24+/DE-30+ or AS5300 T1/E1 CAS - 2600/3600/7200 MTP Required for IOS Gateways - Min 12.0(7)T = Amteva Gate Server Message Store Directory Isolated Deployments

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Router/GW IP WAN (Primary Voice Path) PSTN (Secondary Voice Path) V Site A Site B Site C IP WAN Router IP WAN Router Call Manager V V Call Manager IOS Gatekeeper for Admission Control Call Manager at Site 200 IP Phones 100 Voice Mail Boxes IOS Gateways Required ALL G.711/80kbps for ALL IP WAN voice calls Maximum 10 IP WAN sites Transparent PSTN Fallback if WAN Unavailable No Voice Mail Networking Multi-site WAN Distributed Call Processing

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Router/GW IP WAN Centralized Call Manager PSTN V Location Hub Location 2 Location 3 V IP WAN Router V IP WAN Router Multi-site WAN Centralized Call Processing Centralized Dial Plan 200 IP Phones 100 Voice Mail Boxes Admission Control - Location BW Limits must Remotes must use Selsius GWs due to MTP 128kbps Minimum (Plan for 80kbps per Call even with Selsius Gateways) No Service if WAN down (Unless Dial Backup)

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Hub and Spoke Topologies Admission Control + Capacity Planning Ensure voice traffic at every site does not exceed configured WAN Bandwidth Minimum requirements for Voice, Video and Data shall not exceed 75% of link or VC bandwidth (Remaining 25% for Routing Protocol updates and link layer header BW consumption etc.) Required Link Capacity = (Min BW for Voice + Min BW for Video + Min BW for Data) / 0.75 Multi-site WAN Topologies

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Call Manager 2.4 (Required) Voice/Email Messaging Server uOne GateServer (4.1E) Message Store/Directory Directory uOne 4.1E 1. 100 Voice mail Boxes or less 2. Voice Messaging only 3. Appserver requires Out of Band DTMF and is G.711 only (12.0(7)T) 4. Out of Band DTMF from IOS GW need to be dtmf-relay h245-alphanumeric All components on MCS-7830 Skinny Station Protocol Gateserver registers as an IP Phone MCS-7830 Voice Messaging All in one Entry Package

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uOne Entry Edition Includes 100 user mailboxes 30 minutes storage max per user 4 simultaneous streams (ports) of voice messaging OPTIONAL 4 stream (port) upgrade (Maximum is 8 Ports) To go beyond 8 ports or 100 users you will have to upgrade to a stand alone uOne Enterprise Version

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PSTN V Site A Site B Site C V V A A A CM 3.0 Isolated Deployments Call Manager Cluster at each site 10,000 IP Phones Per Cluster 2500 per Cal Manager No Limit to Number of Clusters (Each Cluster is isolated PBX) G.711 - Makes no sense to compress calls GW Selection based on PSTN signaling MTP Not Required

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PSTN V Site A Site B Site C V V A A A IP WAN (Primary Voice Path) Gatekeeper CM 3.0 Multi-Site Wan Distributed Call processing Call Manager Cluster per site 10,000 IP Phones Per Cluster Max 10 Clusters (5 Redundant) - IOS GK for CAC G.711 or G.729 for WAN (C6K DSP Farm required for G.729 & Conference) GW Selection based on PSTN signaling MTP Not Required

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Location Based Admission Control and Cluster Interaction Not Supported Supported Cluster1CMA Cluster1CME Cluster1CMD Cluster1CMC Cluster1CMB Location 1 Location 2 One CCM in a clusterOR up to Three CCMs in a Cluster Providing: 1. All Phones always registered with same Cisco CallManager 2. Achieved by keeping all Phones with same CM Group list order 3. Creation of a Locations Cluster of two or three 4. Possible Location status synch during CM failover - self healing B A Location 1 Location 2 Primary CM Backup CM Locations WAN Cluster Remote Sites Central Site Remote Sites Central Site C Backup CM

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Router/GW Call Manager IP WAN Compressed Call Leg in the IP WAN DSP Resource at Central Site uOne Gateserver DSP = DSP Farm G.711 Call Leg Compressed Call Leg Supported/Committed for 3.0 DSP 3.0 Solution Set No DSP Farm or Cisco CallManager at remote sites

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GW Router/GW IP WAN Centralized Call Manager V PSTN V Site1 Site2 Dial 9 for Local PSTN GW Partition 1. Devices with similar reachability characteristics 2. Items placed in Partition: IP Phones, DNs, Gateways + Route Patterns Calling Search Space 1. Which Partitions a device may search in for a dialed number 2. Provides dialing permissions/restrictions 3. Each device assigned a Calling Search Space 3.0 Dial Plan Enhancements - Partitions Centralized Call Manager but Distributed Dial Plan

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3600 IP WAN 337-1XXX 447-2XXX V V Dial Plan - Robust Digit Manipulation PSTN Voice Overflow User dials 447-2222 1111 2222 2222 Presented to Call Manager 3600 IP WAN 337-1XXX 447-2XXX V V PSTN Voice Overflow User dials 447-2222 1111 2222 2222 Presented to Call Manager Call Manager strips 447 for IP WAN Call Manager pre-pends 1610 for PSTN Call Placed Across IP WAN Call rejected Across IP WAN, takes PSTN Gatekeeper

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IOS GW Call Manager with Amteva Gateserver 1. Allows for IOS GW/Call Manager Integration without MTP (Greater Scalability) 2.MGCP support for Analog IOS GW interfaces ONLY (FXS & FXO) 2. MGCP support for Analog IOS GW interfaces ONLY (FXS & FXO) 3. Fax will require G.711 Operation with MGCP PSTN V Design Characteristics IOS GW Call Manager with Amteva Gateserver PSTN V Initial Call - Direct from GW to IP Phone No MTP Required Call Transfer - Allow Supplementary Services Skinny Station MGCP Voice Path MGCP Elimination of MTP

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IOS GW Cisco CallManager with Amteva Gateserver 1. Allows for IOS GW/Call Manager Integration without MTP (Greater Scalability) 2. IOS gateways with Digital Interfaces without MTP 3. Minimum IOS 12.0(7)T PSTN V Design Characteristics IOS GW Cisco CallManager with Amteva Gateserver PSTN V Initial Call - Direct from GW to IP Phone No MTP Required Call Transfer - Allow Supplementary Services Skinny Station H323v2 Voice Path Elimination of MTP H.323v2 with Open Closed Logical Channel

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IOS Gatekeeper V V Router/GW CallManager Cluster CallManager Cluster IP WAN Design Characteristics Compressed WAN Call Leg across IP WAN Gatekeeper used for Admission Control with PSTN Fallback Cisco uOne Gateserver Cisco uOne Gateserver V = DSP Farm G.711 Call Leg Compressed Call Leg Site A Site B Call Manager and DSP Farm at each site

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Summary & Conclusion LAN & WAN Infrastructure needs QoS Without it give up and go home ! Network Design should support redundancy Voice systems do today ! Ciscos has solutions to support Voice/Video/Data enabled networks Cisco AVVID Packet Voice Is A Real Solution ! Cisco AVVID Architecture Powerful, flexible IP architecture that supports Intelligent Integrated Applications Cisco AVVID Advanced Applications