VOIP

This EAL-SK200/EAL-SK400/EAL-SK800 series Gatekeeper is designed as an embedded-based platform. It performs internet telephony services for status of registration, call detail record and security registration policy. EAL-SK200/EAL-SK400/EAL-SK800 series Gatekeeper provides centralized address translation and controls access to the H.323 terminal Networks compatible with all H.323 v2/v3/v4 IP telephony products, including the entire Internet Voice Gateway line. Beside of Gatekeeper function, EAL-SK200/EAL-SK400/EAL-SK800 series also build in 2/4/8 voice channels which supports H.323 signal protocols.

	200 H.3232 endpoints scale: SK400 providing 250 H.323 endpoints to register.
	Register Security Policy: SK400 providing security setting on your H.323 VoIP network. This provides protection for VoIP calls and insures proper endpoint identification.
	Pre-Granted Endpoints: letting other gateways or H.323 endpoints which were not register to this embedded gatekeeper. And the registered VoIP Gateways can make an Off-Net call to these pre-granted endpoints.
	Real time Call Detail Record and Post Call Detail Record Report : Support Real Time CDR to monitor VoIP calls, including caller ip, called ip , call date , call duration and other information. Also providing a CDR report to look up VoIP call record.
	Top 20 list: SK400 Gatekeeper can lists top 20 calls by call duration, caller number, calling number, caller IP or callee IP address.
	Syslog Client: Providing CDR information to Syslog Server.

The following application was use in Multi-Office as an example:

	Voice Codec: G.711(A-law /¦Ì-law),G.729 AB, G.723 (6.3 Kbps / 5.3Kbps)
	FAX support : T.30 / T.38
	Echo Cancellation: G.165/G168
	FXO Caller ID detection : DTMF and FSK (Optional)
	FXO hang up detection / anti-seized : Tone Learning Automatically / Manual Tone Learning (Optional)
	Answer supervision: Support Battery Reverse Detection and Voice Detection.
	FXO answer delay time: Support delay 0 ¨C 8000 ms to answer.
	Adjustable AC Termination Impedance : 600 / 900 OHM and complex Impedance
	Failsafe Mechanism (FXS relay to FXO) : Power failed by pass support / Internet Failed by pass (Optional))
	12K Hz and 16K Hz Metering  (Customized)

	H.323 v2/v3/v4
	LAN :NAT, Virtual Server, DHCP Server
	WAN: PPPoE client, DHCP client, Fix IP Address, DDNS client
	Network Address Translation: Providing build-in NAT router function
	Smart QoS: Guarantee the voice bandwidth

	Voice channels status display
	Adjustable volume : - 9 db ~ 9 db
	Silence Compression
	Auto Dial for speed
	Dynamic Jitter Buffer

	web-based Graphical User Interface
	Remote management over the IP Network
	FTP firmware upgrade
	Backup and Restore Configuration file
	Syslog client support

	AC power : AC100V-240V, DC12V/1.5A,50/60 Hz
	Temperature: 0¡ãC ~ 40¡ãC (Operation)
	Humidity: up to 90% non-condensing
	Emission: FCC Part 15 Class B, CE Mark
	Dimension : 260 x 130 x 35 mm
	Weight: 900g (Aluminum)

SK200/SK400/SK800 VoIP Gateway Catalogue

Model: EAL-S200 & EAL-S400 series

The EAL-S200/EAL-S400 Series VoIP Gateway is fully both SIP and H.323 standard compliant residential gateway that provides a total solution for integrating voice-data network and PSTN. By simple installation, this revolutionary compact voice over IP (VoIP) gateway could be configured as a 2/4 port FXS/FXO VoIP Gateway which provides voice connectivity over the IP network and to the Public Switched Telephone Network (PSTN). The Gateway is equipped with a four port Ethernet switch and built-in NAT router function that provides Internet access using only one IP address.

Besides, it provides high voice quality and optimized packet voice streaming over managed and public (Internet) IP networks.

	Both support SIP and H.323 protocols: SIP Registration and Digest Authentication; H.323 Gatekeeper Registration.
	Single Number / Account for multiple ports.
	Caller ID Delivery and Detection: FXS support DTMF&FSK Caller ID generation; FXO supports DTMF&FSK Caller ID detection. (Optional)
	Flexible and Smart VoIP call Dialing Book: VoIP call Book could provide any application VoIP call to any type destination (Domain name / IP address, PSTN or PBX) or hunting number setting.
	AC termination Impedance : 600/900 OHM and complex impedance
	Answer Supervision for Polarity Reversal Detection and Voice detection
	NAT traversal: This feature allow gateway to operate behind any NAT/Firewall device. Need not to change any configuration of NAT/Firewall like setting virtual server.
	Smart-QoS: This feature provides good voice quality when user place a VoIP call and also access internet at the same time. The gateway will automatically start to reserve bandwidth for voice traffic when VoIP call proceeds.
	Call Hunting Facility: This function helps gateway to use the lines effectively. This facility automatically transfers your incoming call to a free line. Subscribers need not indicate numerous numbers of each port of gateway.
	Voice channels status display: This function display each port status like as on-hook, off-hook, calling number called number, talk duration, codec.
	Pulse Dial support: Support pulse dialing generation and detection. (Optional)
	Flash Detection and Generation Program: FXO support Flash Generation and FXS support Flash Detection.

The following application was use in Multi-Office as an example:

	Voice Codec: G.711(A-law /¦Ì-law), G.729 AB, G.723 (6.3 Kbps / 5.3Kbps)
	FAX support : T.30 / T.38
	Echo Cancellation: G.165/G168
	FXO Caller ID detection : DTMF and FSK (Optional)
	FXO hang up detection / anti-seized : Tone Learning Automatically / Manual Tone Learning (Optional)
	Answer supervision: Support Battery Reverse Detection and Voice Detection.
	FXO answer delay time: Support delay 0 ¨C 8000 ms to answer.
	Adjustable AC Termination Impedance : 600 / 900 OHM and complex Impedance
	Failsafe Mechanism (FXS relay to FXO) : Power failed by pass support / Internet Failed by pass (Optional))
	12K Hz and 16K Hz Metering  (Customized)

	H.323 v2/v3/v4 and SIP (RFC 3261) , SDP (RFC 2327), Symmetric RTP, STUN (RFC3489), ENUM (RFC 2916), RTP Payload for DTMF Digits (RFC2833), Outbound Proxy Support.
	LAN :Support Virtual Server, DHCP Server
	WAN: Support PPPoE client, DHCP client, Fix IP Address, DDNS client
	Network Address Translation: Providing build-in NAT router function.
	Smart QoS: Guarantee the voice bandwidth
	IP TOS (IP Precedence) / DiffServ

	Voice channels status display
	Direct Dialing Mode : peer to peer call (support IP Address Call or Domain Name Call)
	Register Call Mode : register to SIP Proxy Server or H.323 Gatekeeper
	Adjustable volume : - 9 db ~ 9 db
	Silence Compression / VAD
	Auto Dial for speed
	Dynamic Jitter Buffer
	Hot-Line Support

	Web-based Graphical User Interface
	Remote management over the IP Network
	FTP firmware upgrade
	Backup and Restore Configuration file
	Syslog client support

	AC power : AC100V-240V, DC12V/1.5A,50/60 Hz
	Temperature: 0¡ãC ~ 40¡ãC (Operation)
	Humidity: up to 90% non-condensing
	Emission: FCC Part 15 Class B, CE Mark
	Dimension : 260 x 130 x 35 mm
	Weight: 900g (Aluminum)

S400/200 VoIP Gateway Catalogue

 

Model: EAL-SB800/EAL-S1600/S2400 Series

By Modular hardware design EAL-SB800/EAL-S1600/EAL-S2400 series makes any combination of the telephone interface. The EAL-SB800/EAL-S1600/EAL-S2400 Series VoIP Gateway is fully SIP and /H.323 standard compliant high density gateway that provides a total solution for integrating voice-data network and PSTN. By modular installation, this revolutionary compact voice over IP (VoIP) gateway could be easily configured as 8/16/24 channels high density FXS/FXO VoIP Gateway which provides voice connectivity over the IP network and to the Public Switched Telephone Network (PSTN).

	Modular design to accommodate various types of telephony interfaces.
	Both support SIP and H.323 protocols: SIP Registration and Digest Authentication; H.323 Gatekeeper Registration.
	Single Number / Account for multiple ports
	Caller ID Delivery and Detection: FXS support DTMF&FSK Caller ID generation; FXO supports DTMF&FSK Caller ID detection. (Optional)
	Smart VoIP call Dialing Book: VoIP call Book could provide any application VoIP call to any type destination (Domain name / IP address, PSTN or PBX) or hunting number setting.
	AC termination Impedance : 600/900 OHM and complex impedance
	Answer Supervision for Polarity Reversal Detection and Voice detection
	NAT traversal: This feature allow gateway to operate behind any NAT/Firewall device. Need not to change any configuration of NAT/Firewall like setting virtual server.
	Smart-QoS Guaranteed: This bandwidth management feature provide good voice quality when user place a VoIP call and access internet at the same time. The gateway will automatically start to reserve bandwidth for voice traffic when VoIP call proceeds.
	Call Hunting Facility: This function helps gateway to use the lines effectively. This facility automatically transfers your incoming call to a free line. Subscribers need not indicate numerous numbers of each port of gateway.
	Voice channels status display: This function display each port status like as onhook, offhook, calling number callee¡¯s number, talk duration, codec.
	Pulse Dial support: Support pulse dialing generation and detection. (Optional)
	Flash Detection and Generation Program: FXO support Flash Generation and FXS support Flash Detection.

	Voice Codec: G.711(A-law /¦Ì-law), G.729 AB, G.723 (6.3 Kbps / 5.3Kbps)
	FAX support : T.30 / T.38
	Echo Cancellation: G.165/G168
	FXO Caller ID detection : DTMF and FSK (Optional)
	FXO hang up detection / anti-seized : Tone Learning Automatically / Manual Tone Learning (Optional)
	Answer supervision: Support Battery Reverse Detection and Voice Detection.
	FXO answer delay time: Support delay 0 ¨C 8000 ms to answer.
	Adjustable AC Termination Impedance : 600 / 900 OHM and complex Impedance
	Failsafe Mechanism (FXS relay to FXO) : Power failed by pass support / Internet Failed by pass (Optional))
	12K Hz and 16K Hz Metering  (Customized)

	H.323 v2/v3/v4 and SIP (RFC 3261) , SDP (RFC 2327), Symmetric RTP, STUN (RFC3489), ENUM (RFC 2916), RTP Payload for DTMF Digits (RFC2833), Outbound Proxy Support.
	LAN :Support Virtual Server, DHCP Server
	WAN: Support PPPoE client, DHCP client, Fix IP Address, DDNS client
	Network Address Translation: Providing build-in NAT router function.
	Smart QoS: Guarantee the voice bandwidth
	IP TOS (IP Precedence) / DiffServ

	Voice channels status display
	Direct Dialing Mode : peer to peer call (support IP Address Call or Domain Name Call)
	Register Call Mode : register to SIP Proxy Server or H.323 Gatekeeper
	Adjustable volume : - 9 db ~ 9 db
	Silence Compression / VAD
	Auto Dial for speed
	Dynamic Jitter Buffer
	Hot-Line Support

	Web-based Graphical User Interface
	RS232 for configuration
	Remote management over the IP Network
	FTP firmware upgrade
	Backup and Restore Configuration file
	Front LCD Panel for System Status and Management
	Syslog client support

	Standard 50 pin RJ-21 Telco connectors
	AC power : AC100V-240V, DC12V/1.5A,50/60 Hz
	Temperature: 0¡ãC ~ 40¡ãC (Operation)
	Humidity: up to 90% non-condensing
	Emission: FCC Part 15 Class B, CE Mark
	Dimension : 440 x 250 x 45 mm
	Weight: 3500g (Aluminum)

 SB800/S1600/S2400 Series VoIP Gateway Catalogue

¡¡

SB800

SB808

SB804

S1600

S1608

S1616

S2400

S2412

S2424

Ethernet Interface

1 WAN + 1 LAN

RS232 Console

1 RS232 console for configuration

LCD Panel

Front LCD Panel for System Status and Management

Telephony Interface

Standard 50-pin Telco connector

FXS/FXO

8 FXS

8 FXO

4FXS+4FXO

16 FXS

8FXS+8FXO

16 FXO

24 FXS

12 FXS + 12 FXO

24 FXO

Modules Combination

SB800

SB808

SB804

1SB800+ 1SM800

1 SB804+     1 SM804

1SB808 + 1 SM808

1SB800+2 SM800

1 SB804+ 2SM804

1 SB808 +2SM808

H.323 v2/v3/v4

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

SIP (RFC3261)

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

Smart-Qos

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

NAT Traversal

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

Echo Cancellation

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

G.711 a / ¦Ìlaw G.723, G729AB

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

T.38/T.30

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

¨¹

 

SB800/1600/S2400 Series VoIP Modules Card Catalogue

Model: EAL-S200/EAL-S400/EAL-S800/EAL-S1600/EAL-S2400 VoIP Gateway Series

The EAL-S200/EAL-S400/EAL-S800/EAL-S1600/EAL-S2400 VoIP Gateway is fully both SIP and H.323 standard compliant residential gateway that provides a Payphone solution for integrating voice-data network and payphone. By simple installation, this revolutionary compact voice over IP (VoIP) gateway could be configured as a 2/4/8/16/24 port FXS VoIP Gateway which provides voice connectivity over the IP network and to Payphone system

Soundwin VoIP Payphone Solution

	User place a call
	The VoIP gateway send an Access-Request to the rate server.
	The rate server reply with an Access-Accept including the meter pulse pattern.
	The gateway decode the meter pulse pattern and make the outbound call on the H.323 / SIP Network.
	After receipt of the Connect message, the gateway begin to apply the meter pulse pattern to send pulse depend on the pulse pattern.

	Both support SIP and H.323 protocols: SIP Registration and Digest Authentication and H.323 Gatekeeper Registration.
	12K Hz and 16K Hz Metering (Customized)
	Radius AAA Client  (Customized)
	Single Number / Account for multiple ports.
	Caller ID Delivery: FXS support DTMF&FSK Caller ID generation;
	Smart VoIP call Dialing Book: VoIP call Book could provide any application VoIP call to any type destination (Domain name / IP address, PSTN or PBX) or hunting number setting.
	Answer Supervision for Polarity Reversal Detection and Voice detection
	NAT traversal: This feature allow gateway to operate behind any NAT/Firewall device. Need not to change any configuration of NAT/Firewall like setting virtual server.
	NAT traversal: This feature allow gateway to operate behind any NAT/Firewall device. There is no need to change any configuration of NAT/Firewall like setting virtual server.
	Smart-QoS: This feature provides good voice quality when user place a VoIP call and access internet at the same time. The gateway will automatically start to reserve bandwidth for voice traffic when VoIP call proceeds
	Call Hunting Facility: This function helps gateway to use the lines effectively. This facility automatically transfers your incoming call to a free line. Subscribers need not indicate numerous numbers of each port of gateway.
	Voice channels status display: This function display each port status like as on-hook, off-hook, calling number callee¡¯s number, talk duration, codec.

	Voice Codec: G.711(A-law /¦Ì-law), G.729 AB, G.723 (6.3 Kbps / 5.3Kbps)
	FAX support : T.30 / T.38
	FXS Battery Revere Generation
	12K Hz and 16K Hz Metering  (Customized)

	H.323 v2/v3/v4 and SIP (RFC 3261) , SDP (RFC 2327), Symmetric RTP, STUN (RFC3489), ENUM (RFC 2916), RTP Payload for DTMF Digits (RFC2833), Outbound Proxy Support.
	LAN :Support Virtual Server, DHCP Server
	WAN: Support PPPoE client, DHCP client, Fix IP Address, DDNS client
	Network Address Translation: Providing build-in NAT router function.
	Smart QoS: Guarantee the voice bandwidth
	IP TOS (IP Precedence) / DiffServ

	Voice channels status display
	Direct Dialing Mode : peer to peer call (support IP Address Call or Domain Name Call)
	Register Call Mode : register to SIP Proxy Server or H.323 Gatekeeper
	Adjustable volume : - 9 db ~ 9 db
	Silence Compression / VAD
	Auto Dial for speed
	Dynamic Jitter Buffer
	Hot-Line Support

	Web-based Graphical User Interface
	Remote management over the IP Network
	FTP firmware upgrade
	Backup and Restore Configuration file
	Syslog client support

	AC power : AC100V-240V, DC12V/1.5A,50/60 Hz
	Temperature: 0¡ãC ~ 40¡ãC (Operation)
	Humidity: up to 90% non-condensing
	Emission: FCC Part 15 Class B, CE Mark

S200/S400/S800/S1600/S2400 VoIP Gateway Catalogue

Model: EAL-S100 / EAL-S102 VoIP Phone

The EALCanada's VoIP Phone supported SIP protocol and equipped with an integrated FXO port for PSTN connectivity. This feature provides user can transfer to answer call from PSTN when using IP call. This unique feature lets user never missed any call from IP calls or PSTN calls especially for business call or emergency call. Otherwise the S102¡¯s FXO interface supports operation of termination and origination by PSTN line. The IP Phone provides high voice quality and optimized packet voice streaming over IP networks.

 

S100	S102	Key Features
	*	Answer PSTN Call:  Holding VoIP call and answer the PSTN call.
	*	Manual Select Call Route: User can select call route by VoIP or PSTN route.
	*	Life Line support: IP Phone switch to PSTN call automatically when Power is off.
*	*	Call History (Incoming calls / Outgoing calls / Missing calls), Phone Book

Application Scenario

The following application was use in Multi-Office as an example:

IP Specifications

	SIP (RFC 3261) , SDP (RFC 2327), Symmetric RTP, STUN (RFC3489), ENUM (RFC2916),RTP Payload for DTMF Digits(RFC2833), Outbound Proxy Support.
	WAN: PPPoE client, DHCP client, Fix IP Address, Dynamic DNS client

Call Features

	Provides Essential Features: Phone Book, Call History (Incoming calls / Outgoing calls / Missing calls)
	Hand free Speaker Phone
	Speed Dialing
	IP address Call / Domain Name Call
	Voice Codec : G.711(a-law /¦Ì -law ), G.723(6.3Kbps/5.3Kbps), G.729AB (8Kbps)

Configuration & Management

	Key and LCD configuration (2x16 characters LCD)
	Web-based Graphical User Interface
	Remote management over the IP Network
	FTP firmware upgrade

General Specification

	AC power : AC100V-240V, DC12V/1.5A,50/60 Hz
	Temperature: 0¡ãC ~ 40¡ãC (Operation)
	Humidity: up to 90% non-condensing
	Emission: FCC Part 15 Class B, CE Mark
	Weight: 735g

S100 Series VoIP Phone Catalogue

	S100	S102
Ethernet Interface	1 WAN + 1 PC
FXO port		1
SIP(RFC3261)	*	*
STUN Client	*	*
Symmetric RTP	*	*
Outbound Proxy Server	*	*
Call History		*
Phone Book	*	*
Life Line		*