Advanced Telecommunication Wireless VOIP Project by Alcon C4U

2011.11.21by ALCON Telecom Taiwan

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Project Briefing

Chittagong Port: A area of 50 sq.km (10km x 5km), contain 110 buildings, and 2000 telephone user including office and residence. The project will establish an advance and independent telecommunication System for the group user, and it will be easy and convenient, and cost effective solutions, also expandable with the population growing.

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Function & Requirement

1.Wireless System for Fast deployment: The latest wireless technology will be deployed instead of cabling2.VoIP System: Using the advance VoIP system, it provide you flexible options from it function, excellent voice quality 3.Cost and time saving: The system is easy expandable & system upgrading4.Easy Interface with Local system: The system can be easily connect to the existing local telecom network

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Project Design

1.Establish Independent High Speed backhaul Network: The Network is design to link between each local Sub-station and central server station for transmitting multi phone lines service

2.Create Wireless Voice link between Each Building & Sub-station: Professional wireless bridge can connect each users telecom service to sub-station

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Project Design

3. Sub-Station: The repeating point between Central server and users. To ensure the high quality transmitting quality4. Central server Center: the head of the system that provide the number and service to each telephone users5. User Site: On the proper location of each building, we will establish a wireless terminal linking to the sub-station. From the terminal, the network cable will reach each office and residence with IP telephone service

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Map of Chittagong Port

6Project coverage area

10km

5km

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High Speed Backhaul Network

7Backhaul Network Central Server Sub Stationwww.cordless4u.com

Wireless Voice Link

8Central Server Sub Station Clients Buildingwww.cordless4u.com

Benefit of ALCON Wireless System

1. Fast Deployment 2. Range expandable up to 100 sqkm3. Flexible of IP Phone location changing4. Clear Voice transmitting5. Sensor /Alarm compatible6. Heavy Duty & All Weather Outdoor Design for

Wireless system7. Solar Power Compatible 8. Independent System without affect by other

network

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Alink-5X FDD

Switching Hub

Alink-5X FDD

(Center Site)

(Sub Station)

(Sub station)VoIP

Switch Center

JPE-5006P

Switching Hub

JPE-5006P

Alink-5X FDD

(Sub Station )

(G1)

(G2)

(G3)

(G4)

JPE-5006P

JPE-5006P

JPE-5006P

JPE-5006P

Building 1IP Phones

Building 2IP Phones

Building 3IP Phones

Building 6IP Phones

Building 7IP Phones

Building 8IP Phones

Building 4IP Phones

Building 5IP Phones

IP Phonewww.cordless4u.com

Equipment Description

• ALINK-5X FDD: Professional 5GHz Super High Capacity Backhaul Bridge System for 20km

• JPE-5006P : Professional 5GHz Full Band Outdoor Bridge System for 3km Range

• VoIP System: Carrier Level VoIP System with 1+1 Hot standby function

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JPE-5006P Professional 5GHz Full Band Outdoor Wireless Bridge

1. Heavy Duty Outdoor Bridge2. Special Design LAN Port for VoIP3. Support Cable up 75 meter Leghth4. Build-in 23dBi Directional Antenna for Long coverage and

stable signal5. Anti-Rust Stainless6. 12~48VDC Multi Voltage compatible

Wireless Equipment Description

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ALINK-5X FDD Professional 5Ghz Super High Capacity Backhaul Bridge System

1. Heavy Duty All Weather Outdoor Design2. Full Duplex transmission to guarantee Voice Quality3. Support multi VoIP channel and auto bandwidth adjust4. Long Range up to 40km with optional dish antenna5. Plug & Play design with signal indicator for easy installing

Wireless Equipment Description

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Equipment Quantity

• Wireless:a) Alink-5X FDD Backhaul Bridge: 30 unitsb) JPE-5006P Wireless Bridge: 180 units

• VoIP System: a) WellSIP6500-HA Proxy (1+1) for 2,000 Users License: 2

Unitsb) WellBilling6600 for 2000 Users License: 2 Unitsc) Database server/DB (RAID5 DISC ARRAY) 1 Unitd) Wellgate5260/4E1/ISDN PRI Trunk Gateway: 1 Unite) Lanphone389S : 2000 Units

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Proposed VoIP-SIP Softswitch + CPE Network Diagram

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L2 SW

PSTN4E1/

120 Ohm

Switch Center

Ethernet

CPE for SubscriberC5 /VoIP-SIP Softswitch

[Active Standby]HA Redundant Proxy ServerWellSIP 6500

Billing ServerWellBilling 6600

Billing Database ServerMS-SQL

Trunk Gateway WellGate 5260

Wireless

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• The Welltech WellSIP 6500 series SIP Telephony Server is the best choice to your convergence VOIP network which covert the requirements from enterprise to service provider. With built-in rich telephony services, WellSIP 6500 enables traditional PABX features to your VOIP convergence platform. Also you can easily upgrade the license or provide the high available service according to the growth of your business without any hardware changes.

WellSIP 6500 Instruction

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WellSIP 6500 Instruction

1. Intelligent Call Routing WellSIP 6500 provides multiple service routing polices to meet different service providers’ requirements (e.g. load balancing, priority, most idle etc.) It enables service provider to tell how to route the call depending on the call results or predefined rules. The incoming prefix match and outgoing prefix insert provides a very easy way to manage your VOIP exchange service.

2. Easy to Configure and Management Full web management interface make you to manage your WellSIP6500 anywhere of the world. You don’t need remember the command lines or operate it on the specified console. Also the system event notice features keep you the system status updated remotely.

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WellSIP 6500 Instruction

3. NAT On-Demanded TraversalDue to the lack of IPV4 address, a lot of customer is using NAT for their network. WellSIP 6500 provides the NAT on-demanded traversal which will only route the voice when needed. It saves the bandwidth and provides better voice quality compare to route each call voice back to server. No CPE modification is required.

4. Voice NAT/Firewall RouterWith built-in SIP and voice routing features, WellSIP 6500 provides a secure and easy way to migrate your Voice IP PBX solution. It acts as a NAT router and firewall role which voice RTP port is only opened when SIP signaling is established successfully.

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WellSIP 6500 Instruction

5. Multiple Access to Receive Calls AnywhereWith provided SIP TCP and UDP protocol, WellSIP 6500 can accept both type of signals and do the conversion when needed. For each protocol, WellSIP 6500 can support up-to 3 service ports which enabling to receive SIP service anywhere of world. Also WellSIP 6500 provides the SIP and RTP packets encryption/decryption features to breakthrough the ISR blocking of your SIP service.

6. High Availability RedundantWellSIP 6500 provides high availability VOIP service by using active and stand-by redundant technologic which provides hot standby and hitless fail-over for stable call to reach mission-critical service requirement. It keeps your service continues running.

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SIP Standard Compliance

• RFC 3261 (SIP) • RFC 2976 (INFO) • RFC 2327 (SDP) • RFC 3262 (PRACK) • RFC 3325 (Private Extensions to SIP) • RFC 3581 (Symmetric Response) • RFC 3264 (Offer Answer) • RFC 3265 (Specific Event Notification) • RFC 3842 (MWI) • RFC 4235 (BLF) • RFC 4028 (Session-Expires) • draft-ietf-sip-cc-transfer-05 • draft-levy-sip-diversion-08 • SIP UDP Transportation • SIP URI

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SIP Registrar Features

• Dynamic Register • Predefine User/NAT User • MD5 Register Authentication • Programmable Nonce Live Time/Valid Period • Programmable General/NAT Max Register TTL • Proxy Peering • Effective/Expired Date • Register Type Selection • High Availability (Redundant) • Register to External Softswitch or Proxy

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SIP Outbound Proxy Features

• Stateful Proxy Server • Call-based RADIUS Authentication • Call-based MD5 Authentication • Sequential Call Forking • Parallel Call Forking • Proxy Peering Support • Hierarchical Outbound Proxy Support • Subscriber/System based Response Timer • Automatic NAT User Detect

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NAT Transversal

• NAT Traversal for Outbound and Inbound • Automatically NAT detection and RTP Proxy • NAT Partition Support • Intelligent RTP Proxy Resource Management

(WS6500/WG5250/RTP server) • External RTP Proxy Resource Support • Preferred NAT Resource Server

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Telephony Features (1)

• Call Transfer/Call Forward • Incoming/Outgoing Number Screening • Group Hunting

–Round Robin, Round Robin (Load Balance) –Priority, Priority (Load Balance) Hunting –Max Idle Time, Max Idle Time (Load Balance) –Ring All with First Ring Back –Ring All with First Answer –ACD Route –Radius Route

• Group Hunting with Dedicated Length• Digit Manipulation for ANI/DNIS • User Group DM • Caller ID Privacy

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Telephony Features (2)

• Call Hold • Call Pickup • Emergency Call • Find Me • ANI Replacement • Non-NAT user only• Short Code • Do Not Disturb • Coloring Ring Back Tone Service • Announcement Service • VMS • Miss Call Notify by Email

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AAA

• Flat CDR File in Local Storage • Real Time RADIUS Billing Message Support • Support Redundant RADIUS Server • Authorization Message for Prepaid Service • Real Time Call Disconnect for Prepaid User • On-Net Call Billing

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Management

• Administrative Log • Auto Daylight Saving • Front Panel LCD Setup • FTP Server—configurable • HTTP server • HTTP SSL support • NTP time synchronization • Password Security (Cannot view via HTML source): max length: 16 • RS232 • System Event Log • Telnet—configurable • Time Zone Support • User Account Manager • Web-based GUI (Chinese/English) • Web-based Real Time Monitor • Support call trace by Easylog 27

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Environmental

• Dimension: 484 × 44 × 450 mm (1U Industrial rack mount) • Weight: 6kg (unit) • Operating Temp. & Humidity

-Temp.: 0°C~55°C (32°F~131°F) -Humidity: 5%~95% RH, non-condensing

• Power Supply: -300W -INPUT: AC 90V~246V auto switch

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• WellBilling 6600 is a high performance, reliabilities and scalable RADIUS billing server. It's flexible rate plan and reseller/subscriber features fulfill the requirements of service provider. With built-in prepaid and postpaid subscriber service, service provider can provide their time to market VOIP service. It is the best shoot of a cost/effective solution.

WellBilling 6600 Overview

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Key Features

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• RADIUS AAA Support • Authentication Message • Authorization Message • Billing Start/Stop Message • RFC 2865, 2866 Compliance with Selected Attributes • Up-to 500k subscribers support • Fully Web Management Interface • Support Prepaid/Postpaid User • Max Call Duration Protection • Immediate Response • Share Secret with MD5 Protection • WellGate 5250 and WellSIP 6500 Full Interoperability • Automatic 6500 provision • Detail Access Log • Provide Basic Report • Support Coin Phone Service • Support Calling ID (ANI) Validation • Support Charge Account • Auto Monthly Charge Deduction

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Key Features

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• Up-to 5 level User Management Administrator Distributor Group Reseller Reseller Subscriber

• Prepaid Service Real Time Balance Deduct Subscriber/Reseller Recharge & Rollback Recharge Log Effective Date/Expired Date PIN Code Generate and Consume

• Postpaid Service Call Detail Record Storage Effective Date/Expired Date CDR Report

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Key Features

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• Flexible Rate Plan Support Up-to 5 Charge Segments per Rate Prefix Effective Date/Expired Date Longest Prefix Match Programmable charge unit, amount and cycle Support Per Call Charge Call Screening Holiday & Night Time Charge Free Monthly Minutes based on Prefix Monthly Free Charge based on Prefix Deductible Monthly Fee

• External Database support MSSQL Built-in DB Connection Pool management

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Subscriber Management

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• Subscriber Management -Subscriber Login -Real Time CDR query -Prepaid/Postpaid User -Subscriber-based Web Access -Rate Group per Subscriber -Recharge Balance per Subscriber -Effective Date/Expired Date

• WellSIP 6500 Integration (Auto Provision)

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Basic Report & Environmental

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• Basic Report -Call History Report -Recharge Report -Subscriber Report -Reseller Report -Prefix Summary Report -Provisional Report -Access Log Report

• Dimension: 484x44x450 mm (1U Industrial rack mount) • Weight: 6kg (unit) • Operating Temp. & Humidity

-Temp.: 0°C~55°C (32°F~131°F) -Humidity: 5%~95% RH, non-condensing

• Power Supply: 250W -INPUT: AC 90V~246V auto switch

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Overview

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• WellGate 5260 is an UNIVERSAL VOIP GATEWAY which navigates the calls in between H.323, SIP and PSTN freely , not simple PSTN to VOIP calls or vice verse. It can easily implement SIP and PSTN, H.323 and PSTN, SIP and H.323, SIP and SIP, H.323 and H.323 calls simultaneously.Withbuilt-in PSTN/VOIP IVR helps service provider to establish their own voice service platform quick. It is suitable for application of Pre-paid Calling card service termination. The built-in both ways IVR provide flexible function for termination and Pin code authentication announcement.

• WellGate 5260 provides an easy to use web-based managing interface. Administrator can use the web based drag and drop call-flow editor to design their VOIP service very quick and without losing any flexibilities. A web-based voice prompt management GUI is also provided to simplify the deployment of IVR related service.

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PSTN Interface

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• Customizable E1/T1 CAS • E1 CAS DTMF - Loop Start FXO Hot-Line • E1 CAS R2 MF • E1 ISDN PRI • ISDN Protocol - I430/I431, Q920, Q930 • PCM law - Alaw/Mulaw selectable • T1 CAS DTMF/R1MF • T1 ISDN PRI • ISDN user side (TE), Network side (NT) • E1 SS7 ISUP • PSTN Line Hunting & PSTN Line Hunting Channel Selection • System Call setup sending time - PSTN (within 3 seconds)/

VOIP(within 2 seconds)

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Key Feature

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• Navigate Call Freely in SIP, H.323 and PSTN • Support SIP RFC 3261 and ITU-T H.323 V5 Simultaneously • Up to 4 Programmable E1/T1 Trunks • PSTN Signaling: ISDN/PRI, CAS, MFC R2, QSIG , SS7 • Support Audio Codec G.711, G.723.1, G.729A, GSM • SIP to PSTN Call and vice versa • H.323 to PSTN Call and vice versa • H.323 to SIP Call and vice versa • SIP to SIP Call • H.323 to H.323 Call • Support up-to 16 Multiple SIP Proxy Servers • Support SIP Proxy, Gatekeeper and P2P Calls Simultaneously • Built-in Universal VOIP Address Book • Support Early Media and SIP Delay Media • Support RADIUS Authentication, Authorization and Accounting• Intelligent PSTN Call Routing and in-Trunk Hunting

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Key Feature

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• Support Flexible VOIP Routing and Account Code • Flexible Digit Manipulation Plan • Support Calling/Called Number Replacement • In-band and out of Band DTMF Transmission • T.38 Fax Relay up to 14400 bps • Dynamic Call Treatment Based on Drag and Drop Call Flow Editor • Built-in PSTN and VOIP IVR • Provides Call Detail Record • Full Web Management Interface • Comply with RoHS• Support up-to 16 SIP Register/Outbound Proxy Servers • Support IP Access Control List with Account Code • Support IP, ANI, DNIS ACL Group • Added VOIP Hunting Features • Support VOIP Hunting based on Priority & Weekday/Time • Support PSTN Line Hunting based on Time • Made in Taiwan

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Management

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• Administrative Log • Auto Daylight Saving • Customizable Time Zone • Front Panel LCD Setup • FTP Server—configurable • HTTP server • HTTP SSL support • Multiple configuration • NTP time synchronization • Password Security (Cannot view via HTML source): max length: 16 • RS232 • System Event Log • Telnet—configurable • Time Zone Support • User Account Manager • Web-based GUI • Web-based Real Time Monitor • Web-based Voice File Management • Telnet, SNMP, FTP configurable: On/Off • System startup time: Less than 2 mins

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Environmental

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• Dimension: 484 × 44 × 450 mm (1U Industrial rack mount) • Weight: 12kg (unit) • Operating Temp. & Humidity

-Temp.: 0°C~55°C (32°F~131°F) -Humidity: 5%~95% RH, non-condensing

• Power Supply: -250W -INPUT: AC 90V~246V auto switch

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IP Phone389S Instruction

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• IPV6 and IPv4 Dual Stack simultaneously • Support up-to 3 Register Accounts at three SIP Servers • Rich Telephony Features • Support Receive Instant Message ( IM ) • Intelligent Phone Book Name Dialing • Dual Flash Memory image backup • Auto Provision feature

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Interface

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• Ethernet port (RJ-45, 10/100 base-T) • 1-LAN port, for connecting to switch • 1-PC port for connecting to PC • 10/100 based Switch • PoE (IEEE 802.3af ) at LAN port : LP389A, LP389SA only • Earphone/Microphone Jack (3.5mm) for Headset: LP389S, LP389SA

DO Not Support • Handset Jack (RJ-10) • DC 12V power input Jack • LCD Display • Display Format: 16 Characters (W) x 2 lines(H) • View Size: 64(W) x 17.9 (H) mm • LCD Type: TN • Language Option: English and Chinese

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IP Network connection

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• IPv4 (RFC 791), IPV6(RFC 2460) • IPv6 Auto Configuration (RFC 4862) • MAC Address (IEEE 802.3) • Static IP • DHCP Client (RFC 2131) • PPPoE• DNS Client • TCP/UDP (RFC 793/768) • RTP/RTCP (RFC 1889/1890) • IPV4 ICMP (RFC 792), IPV6 ICMP(RFC 4443) • TFTP Client • VoIP VLAN Support (802.1Q/802.1P) • HTTP/HTTPS Server • QoS Support • Support IPV4 only, IPV6 only or dual stack mode

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SIP Protocol

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• RFC3261 compliance • Support up-to 3 SIP Register Accounts (IPv4 and IPv6 are selectable) • Support Primary and Backup SIP Proxy • SIP UDP Protocol • Support SIP compact Form • SIP Session Timer (RFC 4028) • MD5 Digest Authentication • Reliability of provisional responses PRACK (RFC3262) • Early/Delay media support • Offer/answer (RFC3264) • Message Waiting Indication (RFC3842) • Event Notification (RFC3265) • REFER (RFC3515) • Support DNS SRV to locate SIP Server (RFC 3263) • Support STUN NAT Traversal • Support “rport” parameter (RFC 3581)

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Audio Codec

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• G.711 A-law/μ-law, G.723.1 (6.3K/5.3K) • G.729A, GSM 6.10 (full rate) • Silence Suppression • VAD/CNG • Automatic Gain Control • In-band/out of band DTMF (RFC 4733 (RFC 2833)/SIP INFO) • Adaptive/Configurable Jitter Buffer • Acoustic Echo Cancellation • Volume Adjustment

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Preference

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• Customized Idle Text • Phone Book • Intelligent Phone Book Name Dialing • Clock, Daylight Saving, Call-timer • Call History of Missed, Received and Dialed • Dialing Plan • Digit Manipulation • Selectable Call Progress Tone • Support Personal Music Ring • Support Silence Ring • Auto Answer Mode • Support Auto Priority (Auto Switch) Calling • Support Specified Line Calling

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Call Features

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• Caller ID display or inhibit • Voice Mail with Indication • Speed Dialing • Call Waiting/Switching between Calls • Call Forward (Busy, Unconditional, No Answer) • Blocking of Anonymous Call • Do not disturb • Call Hold • Call Mute • Call Transfer • Incoming Call Blocking List

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Call Features

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• Music-on-hold support (via IPPBX or local) • Conferencing (3-way conference over phone) • Multi-parties conference (via IPPBX) • Distinct Ring between on-net and off-net calls

(compatible with SIPPBX6200S, 6200GS and 6200N only), 6200x. • Call Pickup (via IPPBX) • Call Park/Retrieve (via IPPBX) • Voice Broadcasting (via IPPBX) • Barge-in & Barge-in Allowance List • Redialing/pre-dialing • Hot Line • Support Peer to Peer Dialing • Support Receiving and Display Instant Message (RFC 3428)

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Management

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• SNTP with Daylight Saving • Multilanguage Web User Interface • Administrative Telnet CLI • 3 Levels of User Access Right with Password protection • LCD Administration Login • Provides System Status • Diagnostics (debug through syslog) • Configuration Backup/Restore • Reset to Default • Dual Firmware Image Backup to avoid crashing • Support HTTP provision through MAC address

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Environmental

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• Dimension: 200 × 95 × 223 mm (Desktop) • Weight: 1.17kg (One unit with packing) • Operating Temp. & Humidity • Temp.: 0°C~45°C (32°F~113°F) • Humidity: 10%~90% relative humidity, non-condensing • Power Adaptor: • INPUT: AC100V~240V, 50/60Hz • OUTPUT: DC 12V, 1.0A • Power Consumption of PoE : 4 Watts

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