Converged Network Design Agenda Consolidation Cisco IP Telephony

Converged Network Design

Agenda • • Consolidation Cisco IP Telephony – IP Telephony Components • • • But First …. Quality Of Service (Qo. S) 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

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

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

Cisco AVVID An End-to-End Architecture Clients Applications Infrastructure VICE R E S Cisco IP Fabric Intelligent Network Services SUP POR T Call. Manager Servers Message Servers Softphone Message Telephony Server Application Servers Video Platforms PCs IP Phones Gateway Switch • Distributed • Adaptive Router • Open • Manageable Content Servers Paging Directory Servers

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 – Call. Manager, voice mail, IVR, etc. • Network infrastructure – Routers, switches, wire, WAN services PSTN

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

Open, Standards-Based Application Servers Legacy PBX Cisco Directory Server PSTN Call. Manager Gateway Qo. S Enabled Catalyst Switch IP Cisco IP Phones Voice-Enabled Router

Open System Applications— New World Ecosystem Call Center Applications from Geo. Tel Acquisition Voice Mail/UMS from Amteva Acquisition Application Servers PSTN Meet Me Conference Call. Manager Telekol Picazo Qo. S Enabled Catalyst Switch Gateway IP

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 10 Base. T Hub • Locally powered • Range of Voice services – G. 711 & G. 723. 1

…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 M 2006, M 2008, M 2009, M 2012, M 2216, and M 2616 – Mitel SX-50, SX-2000 Superset 410, 420, 430 (DNIC), and 4000 series – Siemens Hicom 110 E, 150 E, 300 E All Optiset E, – Rolm 9751 RP 200, RP 300, RP 400, and RP 600 – Ericsson MD 110, Business Phone Dialog 3000 Still waiting for BU decision on product

Cisco PSTN Gateways • All Cisco IOS/Cat. OS Platforms – • • • The link from IP to Circuit Switched - H 323 Gateway Both Digital & Analogue support Support for Supplementary Services (Hold, Transfer, Conference, etc) Available today……. PBX P S T N

Cisco Multiservice Gateways Headquarters Scalabiliy and Density DE 30+ Small Branch Enterprise Branch Central Site/ Regional Branch 1750 3810 7507 7206 3660 3640 2600 7513 7204 5300 7505 7202 3620 Performance

Cisco Call Manager • Provides intelligent Call Processing & PBX functionality – Advanced Call Control – Scalable and on open systems – Fault tolerant • Browser Accessed • Standards based – H 323 and MGCP support • Application Integration – Unified Messaging – Call Centres

Quality of Service - What is it? • Qo. S = Preferential treatment • Qo. S 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

So How Much Bandwidth Does A Voice Call Really Need ? • Voice calls without Compressed Real Time Protocol can take up as much as 80 K on a WAN link when using G. 711 • Administrator can defines which CODEC in G. 711 64 k + Header 79. 5 k (on Ethernet) “regions” – Currently no low-bitrate conferencing 8 k + Header 20 k/10 k (on PPP) . 1 G. 729(a) G. 723. 1 6. 3 k/5. 3 kk + Header 18 k/8 k (on PPP)

Vo. IP Low Speed Link (<768 KBPS) Challenges and Solutions Challenge Cisco Solutions Congestion Intelligent Queuing Delay and Delay Jitter WFQ, IP Precedence, RSVP, Priority Queuing Packet Residency Interleaving Slow Link Freeze-out by Large Packets FRF. 12, MLPPP, IP MTU Size Reduction, Faster Link Bandwidth Consumption Compression Header Size on Low Bandwidth Links Codecs, RTP Header Compression, Voice Activity Detection WAN Oversubscription, Bursting Traffic Management Router Traffic Shaping to CIR, High Priority PVC, Data Discard Eligibility

Network Infrastructure - LAN • Switched Ethernet to the desktop – No shared flat networks - PLEASE !! • VLAN support – Voice in one Data in the other. • Power capability required – Phones are powered you know. • Either in-line or from Power cube • Redundant Network Design (to address availability) • Quality of Service – Bandwidth is not just the answer – Via Buffer Management and Classification of traffic

Network Infrastructure - WAN • Redundancy via Routing Protocols – Need to reroute quickly, 3 minutes is too long for voice • Sufficient WAN bandwidth – Whilst data may ride for free it still needs bandwidth • Quality of Service – Without it your are toast – How many of your customers run 100 Mb in both the LAN and the WAN ? Avoiding Loss, Delay, and Delay Variation (Jitter) Call. Manager Router WAN Multilayer Campus Not as Critical “Initially” Must Be Switched Multilayer Campus WAN Edge “A Must” Qo. S Starts in the WAN Backbone “A Must” Often Overlooked or Misunderstood

Policy Networking • Define policies for applications and users • Distribute policy bindings – Qo. S Policy Servers, Security – Network enforcement nodes Telecommuters • Enable integrated control over Campus network resources Mobile Users Qo. S Security Enterprise Policy Branch Offices Partners

Not Committed

Cisco 7960 IP Phone - Needs Call Manager Release 3. 0 • Professional, Manager • High or Busy Telephone Traffic • Six Lines – Mix Directory Numbers or Features • Full Duplex Handsfree • Display Area: – Calling Information, Feature Access Via “Soft Keys, ” Additional Display Area for Value-added Services and Applications • Built-in Headset Connection · • 10/100 Base. T - 3 Port Switch

Cisco IP Softphone • True mobility – Work from home but answer as if you are in the office. – Work from the hotel room ! – Now true desktop convergence • • TAPI Application Can control associated IP Phone Directory Integration Call Manager Release 3. 0(2) – Currently works with 2. 4

Call Manager Release 3. 0 • • Distributed Call Processing (N+1) Enhanced Call. Manager Database Administration MGCP - Phase 1 Distributed Call/Single CDR Online Non-Intrusive Configuration Changes Telephone UI Software Support Partitioning Enhancements – Toll Restriction User Classes – Distributed Region Number Management – Regionalized Gateway & Tail End Hop Off

Call Manager Release 3. 1 • • 802. 1 P/Q & RSVP Additional Telephone Feature software support MGCP- Phase 2 Class of Service- User Logon Class of Service- User Profile download to Telephone upon Login. Dynamic Comfort Noise adjustment to 7960 OA&M Enhancements for SMB Platforms Valet/User Model – Personal Call Flow agent enhancements • E 911 device auto-location and notification – E 911 call control upon disconnect • • Digit Analysis Translation Tables Malicious Call Trace CDR Support

CM 3. 0 Clusters • (N + 1) Scheme that provides – Enhanced Redundancy – Scalability • CM 3. 0 San. Jose. E – Up to 5 CM(s) Per Cluster – 2500 Devices Per CM • 512 M Memory Required • 10, 000 IP Phones Per Cluster – (5 -1) * 2500 = 10, 000 San. Jose. A San Jose Cluster San. Jose. D San. Jose. B San. Jose. C

How Do Cluster Communicate Internally • Database used is SQL 7. 0 (+ SQL 7. 0 SP 1) – 1 Publisher Per Cluster – Remaining CM’s are Subscribers – (N -1) TCP connections • 25 CM’s = 24 Connections – All Configuration changes made on Publisher • Call Manager (Real Time Data) – Fully Meshed. – (N * (N - 1)) TCP connections • 5 CM’s = (5 x 4) = 20 Connections • 25 CM’s = (25 x 24) = 600 Connections – Real Time Data - Phone / Gateway Registrations etc

How Do Clusters Communicate Internally Sequel Database Intra Cluster Signaling = Full Mesh 7830 Publisher 7820 Subscriber

N+1 example • Three devices are homed to San. Jose. D. All nodes in the network are connected and are relaying route and registration information to each other. San. Jose. A San. Jose. E San. Jose. B San. Jose. D San. Jose. C

N+1 example • San. Jose. D Powered off. The devices lose their connection to CM San. Jose. D. San. Jose. A San. Jose. E San. Jose. B X San. Jose. D San. Jose. C

N+1 example • The devices re-home to other call managers, which then replicate new route and registration information to each other. • The devices experience only a brief outage - (TBD) – Calls in progress are dropped. – Gateway calls are dropped. • Since device operation is identical, users may notice that anything happened. San. Jose. A San. Jose. E San. Jose. B X San. Jose. D San. Jose. C

Call Manager Groups • Each device - – IP Phone – Skinny Gateway • Has a prioritized list of up to 3 Call Managers to which it can connect. • This is called a Call. Manager Group • This system is revertive (TBD) Primary Secondary Last Resort

Next Generation IP Phone Line Powered Line Cards - Cisco 7960 • • • 10/100 switch port Web Client IP Prec = 5, DSCP = EF, Cos = 5 802. 1 Q VLAN Support Can reclassify 802. 1 p • Catalyst Ethernet Line Card • Catalyst 3500, 4 K, 6 K – Uses Pins 1, 2, 3, 6 • Cat 5 k Patch Panel – Uses Pins 4, 5, 7, 8 Phone Plugged in Switch detects IP Phone and applies power CDP Transaction between Phone and Switch IP Phone placed in proper VLAN DHCP request and initialization

Catalyst IP Telephony Enhancements • Catalyst 6 XXX switch family blades – 48 port 10/100 switch module with 48 vdc power to phones • High density DSP resource module (transcoding, mixed codec conferencing) • 8 port T 1 PRI ISDN Vo. IP gateway module • 24 port FXS module • Catalyst 4 XXX switch family – 48 port 10/100 switch module with 48 vdc power • Modular Vo. IP gateway with 2 VIC/WIC and one WIC and two fixed FXO/FXS groups (50 pin Telco connector) • Catalyst 3500 T • Stackable switch with 48 vdc power to phones • VG 200 Gateway – IOS Based Gateway only based on 2600 router platform

Cisco AVVID : Basic Understanding

IP to IP Call • 70330 and 70325 have goes off-hook CM tells 70325 “you have when the conversation • 70330 and 70325 have a conversation sends OHextension message to CM a • call 70330” is overwith 70330 and 70325 conversation on how best • notify CM out of dial picture CM tells to give tone-CMdrops tells 70330 IP address CM they are ‘on-hook’ to make the call work no longer involved • 70330 dials 70325 of 70325 PBX P S T N IP IP 70325 IP IP 70330 PBX

IP to PSTN Call CM notifies 70330 the call Based is completed on Dial Plan, through CMIP 70330 goes off-hook • when the conversation of message Gateway determines PSTN which Gateway • address sends 70330 OH to CM ends, and Gateway • notify 70330 contacts and releases involvement is. Gateway tothat act CM which tells toport give dial tone Call. Manager • connection Call parameters set up with CM notifies call gateway 70330 dialsisnumber in over PSTN space IP IP 70330 IP IP PBX P S T N

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

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 27 th September 1999

Virtual Intelligent Assistant If Sister Calls and I Am in a Meeting, Send to Unified Messaging If It’s the Boss, and I’m not in a Meeting, Ring Me but Don’t Call Cell If It’s a Customer, Ring Me Wherever! ng Ri Check My Calendar u. One ng Ri IP Network R Ring ng Ri Cell / PDA

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 1 XXX, 2 XXX, 3 XXX etc – Limited to 10 Site Deployments • Call Manager uses 128 kbps in the ARQ (Only going to use 80 kbps for G. 711) • Call Manager registers EACH GK Controlled H. 323 device as a separate Vo. IP-GW with Gatekeeper

Isolated Deployments Site A Call Manager = Site B Amteva Gate Server Message Store Directory Router/GW Call Manager V V PSTN Call Manager at each site 200 IP Phones 100 Voice Mail Boxes IP WAN Router Call Manager V IP WAN Router GW Selection based on PSTN signaling – – – PRI - DT 24+/DE-30+ or AS 5300 T 1/E 1 CAS - 2600/3600/7200 MTP Required for IOS Gateways - Min 12. 0(7)T Site C

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

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

Multi-site WAN Topologies 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

Voice Messaging “All in one” Entry Package MCS-7830 Message Store/Directory Messaging Server Voice/ Email Directory u. One Gate. Server (4. 1 E) All components on MCS-7830 Skinny Station Protocol Gateserver registers as an IP Phone Call Manager 2. 4 (Required) u. One 4. 1 E 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 h 245 -alphanumeric”

u. One 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 u. One Enterprise Version

CM 3. 0 Isolated Deployments A A V V Site A PSTN Site B • 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) A V • G. 711 - Makes no sense to compress calls • GW Selection based on PSTN signaling – MTP Not Required Site C

CM 3. 0 Multi-Site Wan Distributed Call processing A A Site A PSTN V V Site B IP WAN • Call Manager Cluster per site – – (Primary Voice Path) 10, 000 IP Phones Per Cluster Max 10 Clusters (5 Redundant) - IOS GK for CAC Gatekeeper A V • G. 711 or G. 729 for WAN – (C 6 K DSP Farm required for G. 729 & Conference) • GW Selection based on PSTN signaling – MTP Not Required Site C

Location Based Admission Control and Cluster Interaction One CCM in a cluster…OR up to Three CCM’s in a Cluster Providing: Central Site Cluster 1 CMA Cluster 1 CME Cluster 1 CMB Cluster 1 CMD 1. All Phones always registered with same Cisco Call. Manager 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 Central Site Backup CM Primary CM A Backup CM B C Cluster 1 CMC Remote Sites “Locations WAN Cluster” Remote Sites Location 1 Location 2 Not Supported Location 1 Location 2 Supported

3. 0 Solution Set No DSP Farm or Cisco Call. Manager at remote sites Compressed Call Leg G. 711 Call Leg u. One Gateserver DSP = DSP Farm Supported/Committed for 3. 0 Call Manager IP WAN DSP Router/GW Compressed Call Leg in the IP WAN DSP Resource at Central Site

3. 0 Dial Plan Enhancements Partitions Centralized Call Manager but Distributed Dial Plan Centralized Call Manager Site 1 Router/GW V Site 2 PSTN V GW Dial “ 9” for Local PSTN GW Partition 1. Devices with similar “reachability characteristics” 2. Items placed in Partition: IP Phones, DN’s, Gateways + Route Patterns IP WAN 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

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

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

Elimination of MTP H. 323 v 2 with Open Closed Logical Channel Skinny Station H 323 v 2 Voice Path Cisco Call. Manager with Amteva Gateserver V V PSTN IOS GW Initial Call - Direct from GW to IP Phone No MTP Required PSTN IOS GW Call Transfer - Allow Supplementary Services Design Characteristics 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

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

Summary & Conclusion • LAN & WAN Infrastructure needs Qo. S – Without it give up and go home ! • Network Design should support redundancy – Voice systems do today ! • Cisco’s 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