8. NET and QUINTUM: End to End product portfolio 6 Enterprise AF (2-8 FXS / FXO) BX/DX (2-8 BRI / 1-2 T1/E1) HG Hybrid (1-4 T1/E1) VX 1200 (1-8 T1/E1 ) VX 1800 (4-24 T1/E1) Branch offices Small to Medium
14. NET and QUINTUM: End to End product portfolio 6 Enterprise AF (2-8 FXS / FXO) BX/DX (2-8 BRI / 1-2 T1/E1) HG Hybrid (1-4 T1/E1) VX 1200 (1-8 T1/E1 ) VX 1800 (4-24 T1/E1) Branch offices Small to Medium
15. NET VX series Carrier applications: MOD applications: Enterprise applications for Microsoft UM & UC: VX 1800 VX1800 VX400 VX900 VX 900T VX 1200
32. Tenor VoIP Gateway Configurations VoIP Gateway IP Network Station Side Ports PBX, Keyswitch, Phones VoIP Gateway PSTN IP Network Trunk Side Ports Station or Access Gateway Trunk Gateway
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36. PacketSaver™ Offers Substantial Improvement in Bandwidth Utilization PacketSaver™ packs multiple VoIP packets to the same destination into a single IP packet, thereby reducing the total amount of packet overhead and network congestion.
37. PacketSaver™ Saves Bandwidth! Improves utilization by up to 50% AFTER silence suppression!!! With PacketSaver and Silence Suppression Without PacketSaver With PacketSaver No Silence Suppression
39. OCS environment Mediation server 1 Mediation server 2 ISDN PRI ISDN PRI Example of High-availability SIP SIP Gateway Gateway PSTN Carrier load balancing for inbound traffic OCS load balancing for outbound traffic On PRI failure, Gateway outage or Mediation server issue, only 50% of traffic available
50. US Navy Datacenter VX900 Ship Submarine VX 900 VX900 All nodes have FXS and 4 T1/E1, Up to 100 Secure calls per node EIA-530 – Multiple links EIA-530 @ 2Mbit 4 T1/E1 EIA-530 @2 Mbit T1/E1 is clocked from Serial by VX (Differentiator) VX acts as clocking for whole Ship, IP included Navy IP Network STE Phones PBX PBX
51. VoIP over Satellite for Mobile units IP Phone (SIP) IP Phone (H323) PC PC IP Phone (SIP) IP Phone (H323) PC PC INMARSAT Terminal INMARSAT Terminal Analog Phone Analog Phone Analog Phone Analog Phone FXS FXS ISDN PRI ISDN S/T Dial on demand PPP Dial on demand PPP ISDN S/T IP data IP data IP data IP data IP data Location 1 Location 2 Location 3 IP Network VX Series VX Series VX Series PSTN Network
53. Satellite BW optimization for Alcatel DVB-RCS Satcom POTS Com. Server E1 ABC F link over IP Ethernet Switch Remote sites OXE IPMG Com. Server OXE IP ODU Central Station Ethernet Switch IP Com. Server OXE E1 IPMG Com. Server OXE 1+1 BUC 200W & 1+1 LNB L-band A9780 DVB-RCS Hub A9780 DVB-RCS terminal Antenna 7.6m Antenna 2.4m 10W BUC & LNB Main recorder IP IP PABX Network FH PABX PABX VX900 VX900 MGW PSTN LAN IDU MGW IP router PSTN
56. MOD Peacekeeping Application Coalition Multinational Command Promina 200 Promina 400 PBX High Speed Data Ports Analog Voice <64Kbps Data Ports Existing Satellite Channel Back to Coalition Command NATO Promina Node 4km Digital Microwave . Digital Microwave 8kM Back to Divisional HQ Meg Link Brigade Battalion Battalion Promina 400 US Uplink US Uplink VX900 Voice over IP Using Standard Phones VX900 Promina/BBS/VX In MOD country VX900 Ethernet LAN Battalion Voice over IP Using Standard Phones High Speed Data Ports Compressed Analog Voice <64Kbps Data Ports Promina 200 VX900 Voice over IP Using Standard Phones High Speed Data Ports Compressed Analog Voice <64Kbps Data Ports High Speed Data Ports Compressed Analog Voice <64Kbps Data Ports Promina 200 VX900 Voice over IP Using Standard Phones High Speed Data Ports Compressed Analog Voice <64Kbps Data Ports Promina 400 1 2 3 4 5 6 7 8 9 * 8 # 1 2 3 4 5 6 7 8 9 * 8 # Voice over IP Calls Home 1 2 3 4 5 6 7 8 9 * 8 # 1 2 3 4 5 6 7 8 9 * 8 #
57. NATIONAL BORDER CONTROL NETWORK V35 BSP Trunk Why NET VX and Promina platforms: - BSP protocol efficiency on Satcom (BW saving + QoS) - VX PBX like functionalities, “Data compatibility” with Alcatel PBXs FXS and Secure calls support Satellites Modem VX900 “PBX Like” Up to 4 Sync. data ports LAN Satellite Modem VX900 “PBX Like” Up to 4 Sync. data ports LAN Satellite Modem BSP Trunk Satellites Modem V35 N x E1 QSIG N x E1 QSIG IP connectivity TAs TAs Up to 200 Sync. Data ports Up to 200 Sync. Data ports Up to 50 remote sites Up to 50 remote sites Hub site A Hub site B - Advanced TDM data support on IP - Cost saving compared to other solutions IP Network IP Network Up to 8 Analog Phones SIP Phones Modem Up to 8 Analog Phones SIP Phones Modem Promina800 VX Blade Promina800 VX Blade Existing PBX Network TA TA
61. LOW COST FIXED TO MOBILE SERVICE BSP Trunks BSP Trunks E1s ISDN Billing Server Network Management Carrier Network (Traffic collection) Enterprise SOHO E1s Multi SIM card Server GSM Cell VX900 VX900 E1s Multi SIM card Server GSM Cell VX900 E1s Multi SIM card Server GSM Cell VX900 BSP Trunks Goals Provide enterprise customers with cheap rates for fixed to mobile calls powered by mobile to mobile negotiated call rate. Requests Transport and split traffic geographically over multiple GSM cells, at the cheapest cost possible. Why NET VX platform? BSP: Bandwidth optimization BSP: Advanced voice QoS (not reliant on network) Reliability and rapid deployment of products Flexible re-routing capabilities Service Provider Network French PSTN IP VPN 1 IP VPN 2
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63. MOBILE WORKERS PREMIUM VOICE SERVICE Goals Provide Mobile Workers with ONE single telephone number Ensure full reachability whatever media and/or handset is available for incoming calls Ensure flexible call routing decision (handset availability, time of day, caller Id…etc), dynamically modifiable with Web and/or IVR based user interface Provide cheap call rate for outgoing calls INCOMING CALL Why NET VX platform? Provides on-board advanced routing tables Leverages SIP functionalities (Presence, call-forward…etc.) Application Programming Interface (scripting) interfaces with IVR and external routing platform Full SIP stack (register, challenge…etc.) IVR Service Provider Network French PSTN GSM Network Internet GSM Cell WIFI/SIP GSM Smartphone VX900 SIP phone Soft phone SOHO WIFI HOTSPOT (Airport…) Soft phone WIFI/SIP GSM Smartphone Soft phone SDSL REMOTE SITE (Hotel…) GSM Voice-Mail ENTERPRISE PBX extension PBX Voice-Mail Billing Server Advanced routing Web server Network Management API & IVR
64. SPEAKING CLOCK APPLICATION Goals An incumbent carrier must provide a speaking clock service Why NET VX platform? - VX powerful flexible API with integrated IVR allows an easy development and a rapid deployment with SS7 support, multiple languages, different time zones, call statistics, billing details, extended Web services - Competitive solutions are all based on legacy TDM platform, twice the price ! Comments R.O.I: 6 months !!! Incumbent Carrier PSTN Network Internet VX900 VX900 SS7 SS7 Incumbent Carrier Management network Speaking Clock Public Web server CDR collection & Stats server Management NTP Server API & IVR API & IVR
65. NEW COUNTRY-WIDE DIALING PLAN Incumbent Carrier PSTN Network VX900 SS7 Incumbent Carrier Management network CDR collection & Stats server Management API & IVR Goals An incumbent carrier wants to provide IVR services to ease dialing plan migration (one additional digit for domestic numbers) Why NET VX platform? - VX powerful flexible API with integrated IVR allows an easy development and a rapid deployment with SS7 support, multiple languages, call statistics, billing details - Leverages an already deployed VX solution (speaking clock) Class5 switch 1/ Call to old number 3/ API/IVR provide new number details And propose automatic re-routing to new number 2/ Call (old number) is redirected by SS7 network to VX 4/ Call (new number) is redirected by VX
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67. TDM TO IP MIGRATION FOR 2 nd TIER CARRIER (step2) SS7 ISDN ENTERPRISE ENTERPRISE Single Billing Server Single Management Pre-paid service Advanced routing Web server VX2500 Why NET VX platform? Support of all existing TDM applications with multi-protocol conversion One-box solution (no additional SBC, SIP proxy…etc) Single management, single billing platform VX API allows rapid deployment of new services (pre-paid, advanced routing, credit-control, Web access…etc) ISDN H323/SIP Service Provider Network Internet PSTN Network Carrier Network (Traffic collection) ITSP2 Network Long Distance Carrier 1 CALL SHOPS ITSP1 Network CALL SHOPS API & IVR
68. LONG DISTANCE OVER SATELLITE VX900 VX900 VX900 SS7 ISDN ISDN VX900 BSP Trunk BSP Trunk BSP Trunk BSP Trunks ISDN SIP/H323 Single Billing Server Network Management Goals Provide competitive toll quality voice services over satellite links between African countries and France Why NET VX platform? BSP: Bandwidth optimization + advanced voice QoS over satellite (not reliant to network) One box solution for remote sites, easy deployment without local support Centralized management and billing Service Provider Network TCHAD PSTN Network CAMEROON PSTN Network MALI PSTN Network French PSTN Network Internet ITSP Network
69. 2 nd TIER CARRIER IN MULTIPLE MIDDLE-EAST COUNTRIES BSP Trunk VX2500 SS7 Pre-paid service Advanced routing Credit control Billing Server SS7 Pre-paid service Advanced routing Credit Control Management Billing Server VX900 BSP Trunk BSP Trunk VX900 SS7 Management Billing Server Pre-paid service Advanced routing Credit control VX2500 BSP Trunk Management US ISDN2 Goals A pre-paid calling card distributor for major GSM operators in Middle East decided to develop a new VoIP business (traffic origination & termination) ensuring smooth development of services in the future. Why NET VX platform? Unequalled networking capabilities: BSP protocol, centralized management, remote survivability, SNMP support Toll voice quality (BSP QoS) for long distance & international calls Scripting capabilities: billing, prepaid platform, credit-control, advanced routing decisions SS7, SIP/H323 native support and any to any protocol conversion. KUWAIT PSTN Network USA Network Internet Service Provider Network BAHRAIN PSTN Network JORDAN PSTN Network Service Provider Network Service Provider Network
70. VX in the Microsoft Unified Communications Market
71. VX support for Office Communications Server 2007 Exchange 2007 provides Unified Messaging: v-mail and e-mail converged Unified Communications includes IM, voice, Video conferencing and Unified Messaging with smart presence integration Unified Communications requires interoperability with existing voice systems and networks
78. Active Directory integration VX PBX or IP-PBX ISDN UM 1. ISDN or IP call comes in VX, Call is made to PBX and PBX extension. VX uses AD to determine if call will be sent to UM or not after 30s ( start or not monitoring the call) 3. If call is NOT answered after 30s, VX forwards or not the call to UM Subscriber Access, redirecting number is properly populated 2. Phone rings for 30 seconds PSTN ITSP SIP-Trunk ISDN, QSIG, CAS , H323 or SIP Hosted Voice Services SIP TLS SRTP AD Active Directory Cache ACTIVE DIRECTORY INTEGRATION API
79. More solutions based on AD integration… VX ISDN UM 1. ISDN FAX call for John comes in VX. 3. VX API will send call to UM Subscriber Access and add diversion header in order to reach John’s UMBox. 2. VX API will route call to Exchange, based on dialed number compared to AD cache details (John’s FAX number) ISDN, QSIG, CAS or H323, SIP SIP PSTN VX ISDN UM PSTN SIP SOLUTION FOR FAX SERVICES SOLUTION FOR UMBOX DIRECT ACCESS BY OWNER FAX CALL FOR JOHN PBX or IP-PBX ISDN, QSIG, CAS or H323, SIP PBX or IP-PBX AD Active Directory Cache 4. FAX call will be left in John’s UMBox. CALL TO LISTEN TO VMs 1. ISDN call from any of John’s numbers comes in VX. 3. VX API will change Caller Id to John’s UM extension 2. VX API will route call to UM, based on dialed number (UM Subscriber Access) and Caller Id (John’s mobile) compared to AD cache details 4. John will reach his UMBox without having to provide his UMBox number UM AD Active Directory Cache JOHN’s HOME OFFICE API API JOHN’s GSM PHONE
80. More solutions based on AD integration… (cont.) VX ISDN UM PSTN SIP SINGLE VOICE-MAIL: UMBOX FOR GSM 1. John’s GSM phone rings, no answer, and forward call to UM Subscriber Access number after 30s. 3. redirecting number is correctly populated using AD cache details. 2. VX API will route call to UM Subscriber Access based on: -dialed number (UM Subscriber Access) -Caller Id (Tom) -Diversion header (John) provided by Carrier UM AD Active Directory Cache VX ISDN UM 1. ISDN call comes in VX, Call is made to SIP phone if present in registrar table 3. If call is NOT answered after 30s, VX forwards the call to UM Subscriber Access, redirecting number is correctly populated using AD cache details. 2. VX monitors call, ringing SIP phone for 30 seconds SIP PSTN VOICE-MAIL FOR SIP DEVICES WIFI SIP Phone 4. Voice mail will be left in proper UMBox. Active Directory Cache AD 4. Voice mail will be left in John’s UMBox. TOM’s HOME API JOHN’s GSM PHONE API
81. Support for Voice Mail on any PBX SOLUTION FOR TDM PBX SUPPORTING VOICE-MAIL SOLUTION FOR IP-PBX NOT COMPATIBLE WITH SIP FOR UM VX Legacy PBX ISDN UM 1. ISDN call comes to PBX, call is made to a PBX extension 3. VX forwards the call to UM subscriber access, redirecting number is properly populated 2. Call rings phone for 30s and PBX forwards to VX PSTN QSIG SIP VX IP-PBX ISDN UM 1. ISDN call comes to PBX, call is made to a PBX extension 3. VX forwards the call to UM Subscriber Access, redirecting number is properly populated 2. Call rings phone for 30s and PBX forwards to VX PSTN SIP H323 SIP
82. Support for Voice Mail on any PBX SOLUTION FOR PBX NOT SUPPORTING VOICE-MAIL SOLUTION FOR ITSP AND HOSTED SERVICES INTEGRATION VX Legacy PBX ISDN UM 1. ISDN or IP call comes in VX, Call is made to PBX, and PBX extension 3. If call is NOT answered after 30s, VX forwards the call to UM Subscriber Access, redirecting number is properly populated 2. VX monitors call, ringing phone for 30 seconds ISDN, QSIG, CAS SIP PSTN VX PBX or IP-PBX ISDN UM 1. ISDN or IP call comes in VX, Call is made to PBX, and PBX extension 3. If call is NOT answered after 30s, VX forwards the call to UM subscriber Access, redirecting number is properly populated 2. VX monitors call, ringing phone for 30 seconds PSTN ITSP SIP-Trunk ISDN, QSIG, CAS, H323 or SIP Hosted Voice Services SIP API API
83. PSTN Remote survivability for UM VX PBX ISDN UM ISDN, QSIG, CAS SIP PSTN HEADQUARTER VX PBX ISDN ISDN, QSIG, CAS IP WAN X REMOTE OFFICE 1. Call comes in VX, Call is made to PBX and PBX extension. If not answered, call is transferred to UM Subscriber Access. 2. If IP WAN is down, VX is able to re-route call over PSTN, 3. Call will reach voicemail services and message can be left . API is used to redirect call over PSTN, re-populate calling Id and diversion header field API API
84. ITSP Remote survivability for UM VX PBX SIP trunk UM ISDN, QSIG, CAS SIP Internet telephony service provider HEADQUARTER VX PBX SIP trunk ISDN, QSIG, CAS IP WAN X REMOTE OFFICE 1. Call comes in VX, Call is made to PBX and PBX extension. If not answered, call is transferred to UM Subscriber Access. 2. If IP WAN is down, VX is able to re-route call over SIP-trunk. 3. Call will reach voicemail services and message can be left. API is used to redirect call over PSTN, re-populate calling Id and diversion header field API API
85. Local Voicemail survivability for UM (future) VX PBX ISDN UM ISDN, QSIG, CAS SIP PSTN HEADQUARTER VX PBX ISDN ISDN, QSIG, CAS VOICE-MAIL CACHING USING API AND IVR IP WAN X REMOTE OFFICE 1. Call comes in VX, Call is made to PBX and PBX extension. If not answered, call is transferred to UM Subscriber Access. 2. If IP WAN is down, VX will store Voice-mail message locally. 3. As soon as IP WAN connectivity is restored, VX will transfer Voice-mail message as email to UM.
87. Support for Active Directory VX PBX or IP-PBX ISDN UC 1. ISDN or IP call comes in VX. VX uses AD to determine if call will be sent to PBX or to UC 2. Migrating users from PBX to OCS is now easily done by IT at AD level. PSTN ITSP SIP-Trunk ISDN, QSIG, CAS , H323 or SIP Hosted Voice Services SIP TLS SRTP AD Active Directory Cache Active directory integration EXT 2962 Communicator
88. NET and VX: Enabling the single phone number… VX ISDN UM PSTN SIP ISDN, QSIG, CAS or H323, SIP PBX or IP-PBX 1. ISDN call comes in for John’s office number 5. Or send the call directly to John’s GSM phone via PSTN access 2. VX looks in AD and route call to OCS (John’s Communicator) 6. VX API will monitor the call and transfer it to UM mailbox of John if GSM does not answer UM/UC AD Active Directory Cache EXT 2962 SIP JOHN’s GSM PHONE JOHN’s WiFi/SIP PHONE TOM calls John office number 3. John’s Communicator is set to simultaneous ring to John’s GSM phone 4. Upon request to call John’s GSM, VX will check about Wifi/SIP presence of John’s phone API Communicator
89. Enabling the single phone number…(future) VX ISDN UM/UC 1. ISDN FAX call for John comes in VX, on his office number. 3. After AD lookup, VX API will send call to UM Subscriber Access and add diversion header in order to reach John’s UMBox. 2. VX API detect the CNG tone provided by remote FAX ISDN, QSIG, CAS or H323, SIP SIP PSTN SOLUTION FOR FAX SERVICES FAX CALL FOR JOHN’S OFFCIE NUMBER PBX or IP-PBX AD Active Directory Cache 4. FAX call will be left in John’s UMBox. API
90. VX & AD – Solving Number & Name Presentation! VX Cisco CCM OCS Server / Mediation Server Active Directory Cache John Smith EXT 2344 Calling Number = +15105742344 Calling Name = AD Lookup +15105742344 Retrieve NAME Field AD Return NAME = John Smith Calling Number = +15105742344 Calling Name = John Smith John Smith Calling Communicator
91. Remote survivability scenarios VX PBX ISDN Mediation server ISDN, QSIG, CAS SIP HEADQUARTER VX PBX ISDN ISDN, QSIG, CAS IP WAN X REMOTE OFFICE 1. VX uses LQM (Link Quality Management) to manage OCS presence/availability or cause code failure when placing call to Mediation, as trigger to start re-routing scenario. 2. Call route table provides ability to route to alternate destination (not mediation server, but PBX), and manipulate destination number, like converting in extension number, or enterprise switchboard, local or in headquarter. 3. Note that it’s possible to create a static entry in routing table, or to get info from Active Directory. SIP UC SIP Mediation server SIP Call forward to extension number or company switchboard Extension or switchboard Extension or switchboard PSTN Communicator John Communicator
92. Remote survivability scenarios VX PBX ISDN Mediation server ISDN, QSIG, CAS SIP HEADQUARTER VX PBX ISDN ISDN, QSIG, CAS IP WAN X REMOTE OFFICE 1. VX uses LQM (Link Quality Management) to manage OCS presence/availability or cause code failure when placing call to Mediation, as trigger to start re-routing scenario. 2. Call route table provides ability to route to alternate destination (not mediation server, but PSTN), and manipulate destination number, like converting in mobile number, or analog line or answering machine number. 3. Note that it’s possible to create a static entry in routing table, or to get info from Active Directory. SIP UC SIP Mediation server SIP Call forward to PSTN numbers like mobile phone, analog lines or answering machine Extension or switchboard Extension or switchboard PSTN Communicator John Communicator GSM Phone Answering machine Analog line
93. Remote survivability scenarios VX PBX ISDN Mediation server ISDN, QSIG, CAS SIP HEADQUARTER VX ISDN IP WAN X REMOTE OFFICE 1. VX uses LQM (Link Quality Management) to manage OCS presence/availability or cause code failure when placing call to Mediation, as trigger to start re-routing scenario. 2. Call route table provides ability to route to alternate destination (not mediation server, but SIP registrar table), in order to reach SIP phones used as backup solution. SIP UC SIP Mediation server SIP Call forward to backup SIP phones for small offices (no PBX) SIP Phone SIP Phone SIP Phone John PSTN Communicator John Communicator Registrar
94. Remote survivability scenarios VX PBX ISDN Mediation server ISDN, QSIG, CAS SIP HEADQUARTER VX PBX ISDN ISDN, QSIG, CAS IP WAN X REMOTE OFFICE 1. VX uses LQM (Link Quality Management) to manage OCS presence/availability or cause code failure when placing call to Mediation, as trigger to start re-routing scenario. 2. Call route table provides ability to route to alternate destination (not local mediation server, but SIP registrar table), in order to reach SIP softphones used as backup solution on the user’s PC. SIP UC SIP Mediation server SIP Call forward to backup SIP softphone Extension or switchboard Extension or switchboard SoftPhone John PSTN Communicator John Communicator Registrar
95. Remote survivability scenarios VX PBX ISDN Mediation server ISDN, QSIG, CAS SIP HEADQUARTER VX PBX ISDN ISDN, QSIG, CAS IP WAN X REMOTE OFFICE 1 1. VX uses cause code failure when placing call to ISDN, as trigger to start re-routing scenario. 2. Call route table provides ability to route to alternate destination, in order to reach PSTN via another VX in the network, using IP WAN. SIP UC SIP Call reroute on ISDN failure in multisite environment Extension or switchboard Extension or switchboard Mediation server VX PBX ISDN, QSIG, CAS Extension or switchboard Mediation server SIP SIP SIP SIP SIP SIP ISDN REMOTE OFFICE 2 PSTN
96. Remote survivability scenarios VX PBX ISDN Mediation server ISDN, QSIG, CAS SIP HEADQUARTER VX PBX ISDN ISDN, QSIG, CAS IP WAN REMOTE OFFICE 1 SIP UC SIP Call reroute on IP WAN failure in multisite environment Extension or switchboard Extension or switchboard Mediation server VX PBX ISDN, QSIG, CAS Extension or switchboard Mediation server SIP SIP SIP SIP SIP SIP ISDN REMOTE OFFICE 2 1. VX uses LQM (Link Quality Management) to manage OCS presence/availability or cause code failure when placing call to Mediation, as trigger to start re-routing scenario. 2. Call route table provides ability to route to alternate destination (not local mediation server, but PSTN call), in order to reach a remote Mediation server via remote VX. X PSTN
97. Remote survivability scenarios (future) VX PBX ISDN Mediation server ISDN, QSIG, CAS SIP HEADQUARTER VX PBX ISDN ISDN, QSIG, CAS IP WAN X REMOTE OFFICE 1. VX is used as a “OCS proxy server “for Communicator client. 2. VX uses LQM (Link Quality Management) to manage OCS presence/availability 3. If IP WAN is down, VX detects it (OCS availability), and is able to re-route call over PSTN, or to other local Communicators, thanks to presence in registrar table. SIP UC SIP Mediation server SIP Communicator survivability using VX SIP registrar table PROXY-REGISTRATION PSTN Communicator Communicator Registrar Communicator
100. Thank You Mike Shephard (michael_shephard@net.com) Igor Koshutin (igor_koshutin@quintum.com) All material presented represents the plans of NET, a publicly traded (NYSE) company. All information represented is subject to change, in part or whole, without notice or consequence. No representation made should be construed as a commitment to deliver or a promise of delivery of any feature or set of features. Timeframes presented are estimates and may change at any time.
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