IP Phone Based on SIP/IAX2

Packing:  10pcs/carton
Model NO.:  VoIP Phone UTP1400
Standard:  SIP/IAX2
Productivity:  10, 000 pcs/month
Shipment Terms:  Air freight/Courier/Sea Freight
Trademark:  Uni-Ta
Origin:  China
Usage:  Telephone
Type:  VoIP Products
Style:  Wired
Export Markets:  North America, South America, Eastern Europe, Southeast Asia, Africa, Oceania, Mid East, Eastern Asia, Western Europe

Product Description IP Phone_VoIP Phone_UTP1400 Introduction:
Professional VoIP Phone Based on SIP/IAX2: Stylish and functional in design, the VoIP phone UTP1400, broadly interoperable with SIP/IAX2 platforms and VoIP hardware from major third party vendors, is ideal for a residence or business using a hosted IP telephony service, an IP PBX, or a large scale IP centrex deployment.

The VoIP phone UTP1400 features a full-duplex speakerphone with advanced acoustic echo cancellation, dot-matrix graphic backlit LCD, additional features including 3-way conferencing, call transfer (blind/attended), call forward, call waiting, DND, Voicemail, SMS, customized dial peer, 3 soft keys, as well as DHCP (client/server), NAT traversal (STUN), VLAN (voice VLAN/data VLAN), QoS with diffserv, VPN (L2TP).

By utilizing the cutting-edge quality of service, echo cancellation, comfort noisy generation and voice compensation technology, the VoIP phone UTP1400 can effortlessly provides the excellent voice quality. Meanwhile, the dual 10M/100Mbps auto-sensing Ethernet ports on the IP Phone allow users to install in an existing network location without interfering with desktop PC network connections. The VoIP phone UTP1400 also provides easy configuration thru manual operation (phone keypad and web interfaces) or personalized automated provisioning via central configuration file for mass deployment.

Key Features of the VoIP Phone UTP1400

- Support 2 SIP lines
- SIP supports SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call
- Compatible with IAX2 protocol
- 3-line dot-matrix graphic backlit LCD
- Dual 10/100Mbps Ethernet ports (switched/routed)
- DHCP (client/server), Static IP, PPPoE for xDSL
- Full-duplex speakerphone with advanced acoustic echo cancellation
- Support codec: G.723.1 (5.3k/6.3k), G.729A/B, G.726, G.711(A-law/µ-law), G.722
- Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
- Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168/165), and AGC (Automatic Gain Control)
- Call features: voicemail, SMS, caller ID display or block, conference call, call Forward, call Transfer (blind or attended), call hold, call waiting, paging and intercom, call park/pickup, join call, click to dial, DND, black list, limited list, call history
- Support comprehensive customized dial peer
- Support NAT Traversal (STUN); VLAN (voice VLAN / data VLAN); QoS with diffserv; VPN (L2TP); DMZ; Firewall; DNS relay
- Support automated provisioning through TFTP/TFP/HTTP for mass deployment
- Support management via web interfaces, keypad and telnet

Specifications of the VoIP Phone UTP1400

Product
Description	Professional VoIP Phone Based on SIP/IAX2
Model	UTP1400
Hardware Specification
WAN Port (for connecting to Internet)	1 X 10/100Mpbs RJ45 port
LAN Port (for connecting to PC)	1 X 10/100Mpbs RJ45 port
Speaker	Full-duplex Speakerphone with advanced acoustic echo cancellation
LCD Display	3 lines dot-matrix graphic backlit LCD
Memory	SDRAM: 8M Flash Memory: 2M
Function Keys	14 dedicated function keys (MWI, Phonebook, Hold, Transfer, Speakerphone, Redial, Mute, Call history, Menu, volume control, etc.) 3 soft keys 4 navigation keys
Features & Benefits
Standard	SIP v1 (RFC2543), v2 (RFC 3261) & correlative RFCs Support 2 SIP lines Support IAX2 protocol SIP supports SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call Compatible with Asterisk, Trixbox and other SIP/IAX platforms
Voice Codec	G.711(A-law/ µ -law), G.723.1 (5.3k/6.3k), G.729A/B, G.726, G.722 (wideband)
Voice Standard	DTMF relay: RFC2833, SIP info Auto Gain Control (AGC) G.168/165 compliant line echo cancellation (LEC) Acoustic Echo Cancellation (AEC) Voice Activity Detection (VAD) Comfort Noise Generation (CNG) Adaptive Jitter Buffer
Call Features	Call waiting, call forward, call transfer (blind / attended), call hold, 3-way conference call, paging and intercom, call park/pickup, auto-answer, join call, click to dial Customized dial peer Caller ID display / block DND (do not disturb), Black List, Limited List Support Voicemail, SMS Call Logs: Incoming call, Outgoing call, Missed call (100 entries each) Phonebook:500 entries MWI: Message Waiting Indicator
Network and Management
Access Mode	DHCP (client/server), Static IP, PPPoE for xDSL
Management	Web, Keypad, Telnet management Management with different account right Auto-provisioning through TFTP/FTP/HTTP Firmware upgrade through TFTP/ FTP Configuration file download/upload Support Syslog
Protocols	TCP/IP/UDP, DHCP, PPPoE, SNTP, STUN, MD5, DNS, RTP, RTCP, Telnet, HTTP, FTP, TFTP
Applications	NAT Traversal (STUN); VLAN; QoS with diffserv; VPN (L2TP) ; SRTP security protocol; SNTP Client; DMZ; Firewall; DNS relay; Main DNS and secondary DNS server.
Operating Requirements
Operating Temp.	0~40 degree C
Storage Temp.	-25~60 degree C
Operating Humidity	10~90% Non-condensing
Storage Humidity	10~90% Non-condensing
Power Requirement	Input 100~240V AC, Output 5V DC 1A
Power Consumption	Idle: 1.5W Active: 1.8W
Regulatory Compliance	CE, FCC part 15 class B, RoHS
Packages Contents
UTP1400 IP Phone unit	1
Power Adapter	1
RJ45 Ethernet Cable	1
CD with User Manual	1

Low Cost IP Phone in Promotion (VI2006)

Packing:  10PC/ Carton
Model NO.:  VI2006
Productivity:  10, 000 PCS/ Month
Trademark:  VOPTech or OEM
Origin:  China
Usage:  Telephone
Type:  VoIP Products
Style:  Wired
Country of Origin:  China
Brand Name:  Voptech or OEM
Protocol:  Sip
Export Markets:  North America, South America, Eastern Europe, Southeast Asia, Africa, Oceania, Mid East, Eastern Asia, Western Europe

Product Description SIP Phone designed for service provider
2 SIP lines
2 RJ 45 Ports
Super Low Cost
POE Optional (add 6 USD/ PC)

Basic Features	Support SIP 2.0 (RFC3261) and correlative RFCs
	Support  2 sip line
	Codec: G. 711A/u, G. 7231 high/low, G. 729a/b, G. 722,
	Echo cancellation: G. 168 Compliance in LEC, additional acoustic echo cancellation(AEC) can reach 96ms max filter length in hands-free mode
	Support Voice Gain Setting, VAD, CNG
	Full duplex hands-free speakerphone
	DTMF MODE: DTMF_RELAY, SIP_INFO, RFC_2833
	NAT transverse: support STUN client
Application Features	SIP application: support Call forward/transfer/holding/waiting/3 way call/call park/join call/click to dial
	SIP support SIP domain, SIP authentication (none, basic, MD5), Peer to Peer call
	Call control features: flexible dial map, hotline, empty calling no. reject service, black list for reject authenticated call, white list, limit call, no disturb, caller ID,
	Incoming calls /outgoing calls / missing calls. Each support 100 records
	Support Phonebook 500 records
	10 kind of ring type
	Support Conference
	Support Mute
	Support MWI
Network Features	WAN/LAN: supports Bridge
	Support PPPoE for xDSL
	Support DHCP client on WAN
	Support SNTP Client, Firewall
	Support main DNS and seconary DNS server
	Support VLAN
	QoS with DiffServ
	Network tools in telnet server: including ping, trace route, telnet client
Maintenance and Management	Web, telnet and keypad management
	Management with different account rights
	Upgrade firmware  through POST mode
	Safe mode provide reliability
	Support Auto Provisioning (upgrade firmware through HTTP, FTP or TFTP)
	Support Syslog

Vonets VoIP Gateway With 1WAN*1LAN

Packing:	PE Bag
Standard:	Profile Type
Productivity:	1 Million/Year
Origin:	China
Usage:	Telephone
Type:	Communication Cable
Style:	Wired

Packing:  White Box, Carton Package
Standard:  FCC/CE
Productivity:  5000PCS/Month
Trademark:  VoNets
Origin:  China
Usage:  Telephone
Type:  VoIP Products
Style:  Wired
Size:  74x106x25mm
Color:  Silver
Export Markets:  North America, South America, Eastern Europe, Southeast Asia, Africa, Oceania, Mid East, Eastern Asia, Western Europe

Product Description Product Characteristics

Supports SIP 2.0, TCP/UDP/IP, RTP/RTCP, ICMP, ARP/RARP, NTP, PPPoE, TFTP, HTTP, etc
Supports G. 711 (alaw and u-law), G. 723.1 (5.3K/6.3K), G. 726 (40K/32K/24K/16K), G. 728, G. 729A/B and iLBC...
Built-in Router, NAT, Gateway and DMZ port forwarding. Can also be configured to function as a two Ethernet ports bridge (NAT function is disabled)
Powerful digital signal processing (DSP) to ensure superb audio quality; Advanced adaptive jitter control and packet loss concealment technology
Supports standard voice features such as Call keeping, Call Waiting, Hold, Transfer Forward, FLASH; Supports three way conference
Supports VAD, CNG, AGC, the echo cancellation (G. 168) and Silence suppression.
Supports layer 2 (802.1Q VLAN, 802.1p) and layer 3 QoS (ToS, iffServ, MPLS)
Powerful remote management by TFTP or HTTP mode.
To dispose through the browser or the voice hint.
Supports auto-crossing firewall (NAT);
Supports CALL Progress free enactment function.
Supports the volume adjustment of gateway
Supports SIP encrypts mode, avoid force-out
Supports remote upgrade and upgrade configuration function
Supports passing through and T. 38 fax mode.

SIP 2.0	YES
Router/bridge	YES
DHCP	YES
EthernetPort	1WAN/1LAN(RJ45)
FXS	1FXS
FXO	PSTN pass-through
Size	74X106X25mm
Weight	0.38kg
Humidity	10-90%
Power  supply adapter	AC IN: 100V-240V  DC OUT: +9V/600mA Use power: < 2.5W
Authentication	FCC/CE

10 Pair Connection Module TMTCM+10pair

Product Description

P/N: TMTCM+10PAIR

Description: TeleMax 10 Pair Connection Module

1. Material: Housing-ABS UL94V-2
Base-ABS UL94V-2
Contact -0.5mm thickness
Phosphor bronze plating over nickel over silver
2. Cable: Wire sizes:
Solid copper wires up to 0.4mm-0.8mm in DIA
Insulation PE or PVC insulated wire
3. Wire: AWG 24-26#

Linksys-IP Gateway--SPA2102 (2 FXS, 1 LAN)

Product Description

Include: :
1 - SPA2102 Phone Adapter with Router ----UNLOCKED
1 - 5v Power Adapter ----1 PCS (100V-240V Fress distribution to the joints, European guests with EUR plug, UK guests with UK plug Australia guests with Australia plug) -----Compatible power supply, non-LINKSYS Power.
1 - RJ45 Ethernet Cable
1 - Phone Cable
1 - Quick Installation Guide (Book)
There's no need to replace your analog telephones just because you want to refresh your phone system. Formerly part of the Linksys Voice System (LVS), now part of Cisco Small Business Voice Gateways and ATAs, the SPA2102 phone adapter with router connects standard analog phones or fax machines to an IP-based data network.

With the SPA2102 users are able to protect and extend their investment in telephones, conference speakerphones, and fax machines, as well as control their migration to IP with an affordable and reliable solution. The enhanced and cost-effective benefits of voice over IP (VoIP) service require only a broadband connection to the Internet.

Additional features of the Cisco SPA2102 include:
Dual ports to connect existing analog phones or fax machines, or to a PBX system
Two 100BaseT RJ45 Ethernet interfaces (LAN-WAN) to connect to a home or office LAN, as well as an Ethernet connection for a broadband modem or router (WAN)
SPA2102 Router IP adress: 192.168.0.1
Independent configuration of each line using software controlled by the service provider or the end user
Strong security with reliable, encryption-based methods for communication, provisioning, and servicing

Specifications:
Two voice ports (RJ11) for analog phones or Fax machines
Impedance Agnostics - 8 Configurable Settings
Call Waiting, Cancel Call Waiting, Call Waiting Caller ID
Caller ID with Name/Number (Multi-national Variants)
Caller ID Blocking
Call Forwarding: No answer, Busy, All
Do Not Disturb
Call Transfer
Three-way Conference Calling with Local Mixing
Message Waiting Indication - Visual and Tone Based
Call Return
Call Back on Busy
Call Blocking with Toll Restriction
Delayed Disconnect
Distinctive Ringing - Calling and Called Number
Off-hook Warning Tone
Selective/Anonymous Call Rejection
Hot line and Warm Line Calling
Speed Dialing of 8 Numbers/Addresses
Music on Hold
Fax: G. 711 Pass Through or Real Time Fax over IP via T. 38
Diverse network conditions. Excellent voice quality in a demanding IP network is consistently achieved via our advanced implementation of standard voice coding algorithms. The SPA2102 is interoperable with common telephony equipment like voicemail, Fax, PBX, and interactive voice response systems.
* Large-Scale Deployment and Management
* The SPA2102 offers all the key features and capabilities with which service providers can provide customized services to their subscribers.
* The SPA2102 can be remotely provisioned and supports dynamic, in-service software upgrades. A secure profile upload saves providers the time, expense and hassle of managing and pre-configuring or re-configuring customer premise equipment (CPE) for deployment.
* Ironclad Security
* Linksys understands that security for both end users and service providers is a fundamental requirement for a solid, carrier-grade telephony service. The SPA2102 supports secure, encryption-based methods for communication, provisioning and servicing.
And call routing intelligence programmed into the SPA-3102 NA will connect the caller via the Internet to the far end destination with security and ease. Using the SPA3102 NA at the far end, calls can be answered immediately or further processed as a local call to any legacy land line or mobile phone allowed by the SPA-3102 NA dial plan.

If power is lost to the unit or the VoIP service is down, calls can be sent to a traditional carrier via the FXO interface.

3 Lines Business VoIP Phone With Large Graphic LCD

Packing:	10pcs/carton
Model NO.:	VoIP Phone UTP3000
Standard:	SIP/IAX2
Productivity:	10, 000 pcs/month
Shipment Terms:	Air freight/Courier/Sea Freight
Trademark:	Uni-Ta
Origin:	China
Usage:	Telephone
Type:	VoIP Products
Style:	Wired
Export Markets:	North America, South America, Eastern Europe, Southeast Asia, Africa, Oceania, Mid East, Eastern Asia, Western Europe

Product Description Introduction of the VoIP Phone UTP3000

3 Lines Business VoIP Phone with Large LCD: The UTP3000 is the latest enhanced business VoIP phone of Uni-Ta. With sleek and elegant design, large 128X96 high resolution graphic backlit LCD (3.2 inch), RJ9 and 3.5MM headset jacks, 3 SIP-line keys, 6 programmable keys, 3 dynamic content-sensitive soft keys, a 5-position navigation key, including keys for voice mail, call transfer, call hold, mute, volume control, plus an excellent full-duplex speakerphone with advanced acoustic echo cancellation, make the UTP3000 a full-feature and high quality VoIP phone for enhancing office productivity.

The feature of dual auto-sensing switched/routed Ethernet ports allows the VoIP phone UTP3000 to share internet connection with computers, help to reduce cable clutter and expense. Integrated with IEEE 802.3af Power over Ethernet, the VoIP phone UTP3000 can receive power through Ethernet cable, eliminating the need for external power supply and allows easy deployment with centralized powering and backup. The VoIP phone UTP3000 also supports remote, zero-touch automated provisioning and upgrade from variety severs, such FTP/TFTP/HTTP, good for mass deployment.

The VoIP phone UTP3000 is capable of handling peer-to-peer and SIP proxy / IP PBX registration, authenticated to interact with major IP PBXs / SIP Gateways / VoIP phones in the market. It's the delivery platform for IP voice services that makes benefits from the internet telephony in business communication services.

Key Features of the VoIP Phone UTP3000

- 3 lines indicators with individual SIP account profiles
- SIP supports SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call
- Compatible with IAX2 protocol
- Large 128X96 high resolution graphic backlit LCD (3.2")
- RJ9 and 3.5MM headset jack
- 6 programmable keys, 3 context-sensitive soft keys, a 5-position navigation key, volume keys and predefined keys for voicemail, call transfer, call hold, mute, redial, speaker, phonebook, etc.
- Full-duplex speakerphone with advanced acoustic echo cancellation (96ms max filter length).
- Dual 10/100Mbps Ethernet ports (switched/routed) with integrated Power over Ethernet (802.3af)
- Support DHCP (client/server), Static IP, PPPoE for xDSL
- Support codec: G.711(A-law/u-law), G.723.1 (5.3k/6.3k), G.729A/B, G.726, G.722
- Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
- Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168/165), and AGC (Automatic Gain Control)
- DTMF relay: RFC2833, SIP info
- Call features: voicemail, SMS, caller ID display or block, conference call, call Forward, call Transfer (blind or attended), call hold, call waiting, paging and intercom, call park/pickup, join call, click to dial, DND, black list, limited list, call history
- Call Logs: Incoming call, Outgoing call, Missed call (100 entries each); Phonebook: 500 entries
- Support NAT Traversal (STUN); VLAN (voice VLAN / data VLAN); QoS with diffserv; VPN (L2TP); SRTP security protocol; SNTP Client; DMZ; Firewall; DNS relay; Main DNS and secondary DNS server.
- Support auto-provisioning through TFTP/TFP/HTTP for mass deployment
- Support management via keypad, web interfaces and telnet
- Reversible base stand / wall mount

Specifications of the VoIP Phone UTP3000

Product
Description	3 Lines Business VoIP Phone with Large LCD
Model	UTP3000
Hardware Specification
WAN Port (for connecting to Internet)	1 X 10/100Mpbs RJ45 port Power over Ethernet (802.3af) compliant
LAN Port (for connecting to PC)	1 X 10/100Mpbs RJ45 port
Headset Jack	RJ9 and 3.5MM headset jack
Speaker	Full-duplex Speakerphone advanced acoustic echo cancellation
LCD Display	128X96 high resolution graphic backlit LCD LCD Size: 3.2"
Memory	SDRAM: 16M Flash Memory: 4M
Function Keys	10 dedicated function keys (Voicemail, Phonebook, Hold, Transfer, Speakerphone, Redial, Mute, volume control, RLS ) 3 SIP-line indicators 3 context-sensitive soft keys 6 programmable keys 5 navigation keys
Features & Benefits
Standard	SIP v1 (FRC2543), v2 (RFC3261) Support 3 SIP lines Support IAX2 protocol SIP supports SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call Compatible with Asterisk, Trixbox and other SIP/IAX platforms
Voice Codec	G.711(A-law/ µ -law), G.723.1 (5.3k/6.3k), G.729A/B, G.726, G.722 (wideband)
Voice Standard	DTMF relay: RFC2833, SIP info Auto Gain Control (AGC) G.168/165 compliant line echo cancellation(LEC) Acoustic Echo Cancellation (AEC) Voice Activity Detection (VAD) Comfort Noise Generation (CNG) Adaptive Jitter Buffer
Call Features	Call waiting, call forward, call transfer (blind / attended), call hold, 3-way conference call, paging and intercom, call park/pickup, auto-answer, join call, click to dial Customized dial peer Caller ID display / block DND (do not disturb), Black List, Limited List Support Voicemail, SMS Call Logs: Incoming call, Outgoing call, Missed call (100 entries each) Phonebook:500 entries MWI: Message Waiting Indicator
Network and Management
Access Mode	DHCP (client/server), Static IP, PPPoE for xDSL
Management	Web, Keypad, Telnet management Management with different account right Auto-provisioning through TFTP/FTP/HTTP Firmware upgrade through TFTP/ FTP Configuration file download/upload Support Syslog
Protocols	TCP/IP/UDP, DHCP, PPPoE, SNTP, STUN, MD5, DNS, RTP, RTCP, Telnet, HTTP, FTP, TFTP
Applications	NAT Traversal (STUN); VLAN; QoS with diffserv; VPN (L2TP) ; SRTP security protocol; SNTP Client; DMZ; Firewall; DNS relay; Main DNS and secondary DNS server.
Operating Requirements
Operating Temp.	0~40 degree C
Storage Temp.	-25~60 degree C
Operating Humidity	10~90% Non-condensing
Storage Humidity	10~90% Non-condensing
Power Requirement	Input 100~240V AC, Output 5V DC 1A
Power Consumption	Idle: 2.5W Active: 2.8W
Regulatory Compliance	CE, FCC part 15 class B, RoHS
Packages Contents
UTP3000 IP Phone unit	1
Power Adapter	1
RJ45 Ethernet Cable	1
CD with User Manual	1

10 Pair Pouyet Distribution Box

Product Description 1. The box are suitable for outdoor bad condition, the same as 3M Pouyet type terminal box.
2. The box are dustproof, waterproof, ultraviolet radiation resistant.
3. The inside box serves to install switching equipment.
4. Outside frame of the box provides protection of switching equipment from negative climatic
And also mechanical effects.

Material:
1. Material of Plastic Parts: ABS ultraviolet radiation resistant.
2. Material of metal Parts: Stainless Steel.

Size: 264mm × 240mm × 76mm.
Application:
1. It is intended for switching adjacent cables of main and distributive parts of customer line and line of local telecommunication, carrying out electric measurements and switching telephone circuits.

2. The box can be mounted on STB module ( drop wire connection module), and capability 10 PCS STB modules.

VoIP (PAP2T NA)

Product Description VoIP

1. Linksys VoIP Phone Adapter
2. Two voice ports (RJ-11)
3.1 10BaseT RJ-45 Ethernet Port
4. Session Initiation Protocol v2

VoIP

PAP2T Specifications:

Features

. Two voice ports (RJ-11) for analog phones or Fax machines

. Impedance Agnostics - 8 Configurable Settings

. Call Waiting, Cancel Call Waiting, Call Waiting Caller ID

. Caller ID with Name/Number (Multi-national Variants)

. Caller ID Blocking

. Call Forwarding: No answer, Busy, All

. Do Not Disturb

. Call Transfer

. Three-way Conference Calling with Local Mixing

. Message Waiting Indication - Visual and Tone Based

. Call Return

. Call Back on Busy

. Call Blocking with Toll Restriction

. Delayed Disconnect

. Distinctive Ringing - Calling and Called Number

. Off-hook Warning Tone

. Selective/Anonymous Call Rejection

. Hot line and Warm Line Calling

. Speed Dialing of 8 Numbers/Addresses
Specification:

Data networking:

MAC Address (IEEE 802.3)

IPv4 - Internet Protocol v4 (RFC 791) upgradeable to v6 (RFC 1883)

ARP - Address Resolution Protocol

DNS - A Record (RFC 1706), SRV Record (RFC 2782)

DHCP Client - Dynamic Host Configuration Protocol (RFC 2131)

ICMP - Internet Control Message Protocol (RFC792)

TCP - Transmission Control Protocol (RFC793)

UDP - User Datagram Protocol (RFC768)

RTP - Real Time Protocol (RFC 1889) (RFC 1890)

RTCP - Real Time Control Protocol (RFC 1889)

DiffServ (RFC 2475), Type of Service - TOS (RFC 791/1349)

SNTP - Simple Network Time Protocol (RFC 2030)

Voice Gateway SIPv2:

Sension Initiation Protocol v2 (RFC 3261, 3262, 3263, 3264)

SIP Proxy Redundancy - Dynamic via DNS SRV, A Records

Re-registration with Primary SIP Proxy Server

SIP Support in Network Address Translation Networks - NAT (incl. STUN)

Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP

Codec Name Assignment

Voice Algorithms:

G. 726 (16/24/32/40 kbps),

G. 729 A,

G. 723.1 (6.3 kbps, 5.3 kbps)

Dynamic Payload

Adjustable Audio Frames per Packet

Fax Capability:

Fax Tone Detection Pass-Through

Fax Pass-Though - Using G. 711

DTMF: In-band & Out-of-band (RFC 2833) (SIP Info)

Flexible Dial Plan Support with Interdigit Timers and IP Dialing

Call Progress Tone Generation

Jitter Buffer - Adaptive

Frame Loss Concealment

Full Duplex Audio

Echo Cancellation (G. 165/G. 168)

VAD - Voice Activity Detection with Silence Suppression

Attenuation / Gain Adjustments

Flash Hook Timer

MWI - Message Waiting Indicator Tones

VMWI - via FSK

Polarity Control

Hook Flash Event Signaling

Caller ID Generation (Name & Number) - Bellcore, DTMF, ETSI

Music on Hold Client

Streaming Audio Server - up to 10 sessions

Security

Password Protected System Reset to Factory Default

Password Protected Admin and User Access Authority

Provisioning/Configuration/Authentication:

HTTPS with Factory Installed Client Certificate

HTTP Digest - Encrypted Authentication via MD5 (RFC 1321)

Up to 256-bit AES Encryption

Unlocked Linksys SPA3000 VoIP Gateway Router

Product Description SPA3000

Place of Origin: Guangdong China(Mainland)

Brand name: Linksys

Model Number: SPA3000

Type: VoIP Gateway

LINKSYS SPA3000 SPA-3000 VoIP FXS PSTN-Ulocked
The SPA3000 continues to deliver on Sipura Technologys mission to provide market leading, best-in-class VoIP end points providing fredom and opportunity to service providers and end users. The SPA-3000 features VoIP adapter functionality found in the SPA-2000 and SPA-1000/1 with the additional benefit of an integral connection for legacy telephone network hop-on, hop-off applications. SPA-3000 users will be able to leverage their broadband phone service connections more than ever by automatically routing local calls from cellphones and land lines to a VoIP service provider and vice versa. A typical user clling from a land line or mobilephone will be able to reduce and even eliminate international and long distance telephone charges by first calling their SPA-3000 via a lacal phone number or by using a telephone connected directly to the unit. The advanced authenticacantion and routing intel-ligence programmed into the SPA-3000 will connect the caller via the internet to the far end destination with security and ease. Using the SPA-3000 at the far end, calls can be answered immediately or further processed as a local call to any legacy land line or mobilephone allowed by the SPA-3000 dial plan.
If power and/or IP network connectivity is lost to the unit or VoIP service is down, calls can be sent to a traditional carrier via the FXO interface.

SPA-3000 Telephony Features:
VoIP to PSTN(USA) Service Call Origination and Termination
PSTN(USA) to VoIP Service Call Origination and Termination
Single or Dual Stage Dailing
Service Authentication via PIN, Digest, Caller ID (Bellcore Type 1)
Per Call Authentication and Associated Routing
Least Cost Routing Support
Terminating Impedance Agnostic-8 Configurable Settings
Call Waiting, Cancel Call Waiting
Caller ID Detection(Bellcore Type 1)
Callers ID with Name/Number(Multinational Varients)Display
Caller ID Blocking
Call Waiting Caller ID with Name/Number
Call Forwarding to PSTN or VoIP Service: No Anwser/Busy /All
Do Not Disturb
Call Transfer
Three-way Conference Calling with Local Mixing
Message Waiting Indication-visual and Tone Based
Cll Blocking with Toll Restriction
Distinctive Ringing
Off-Hook Warning Tone
Selective/Anonymous Call Rejection
Hot line and warm line Calling
Speed Dialing of 8 Numbers/Adresses
Music on Hold
Forward Calls to VoIP service -Selective, Authenticated, All
Forward Calls to PSTN service -Selective, Authenticated, All
PSTN line Sharing with Multipal Extensions
Automatic PSTN Fallback(Loss of Power or IP Service to Unit with Quiescence to Normal Operations)
Advanced Inbound and Two Stage Dailing
Independent Configurable Dial Plans-up to 8
Force PSTN Disconnection
Sequential Dailing Support

Packaging Details
Unit Type: Piece
Package Weight: 0.8kg (1.76lb. )
Package Size: 10cm x 10cm x 3cm (3.94in x 3.94in x 1.18in)

Add to Basket

Productivity:	50.000 PCS/Year
Min. Order:	1 Piece
Transportation:	Express Saver, Expedited, EMS, DHL, FedEx, TNT
Type:	VoIP Gateway
Place of Origin:	Taiwan
Brand Name:	Linksys
Model Number:	SPA3000
Export Markets:	North America, South America, Eastern Europe, Southeast Asia, Africa, Oceania, Mid East, Eastern Asia, Western Europe

Three Leg Tube Tower

Product Description

Tube column tower, three leg pipe tower, three leg tube tower
Product name: Communication tower
Function: Signal antenna support equipment
Price $1150 to $1850 per ton
The minimum order 1 set
Product description:
1. Belong to signal a launch, also called signal towers or signal tower
Product description: Into the ground communication detail tower, roof communications tower (also called communications tower) whenever the user choice ground, hills, mountains or roof build towers, all play rack high communication antenna effect;
2. Increase communication or television transmitting service radius, to achieve the ideal professional communication effect.
3. Communication tower is mainly used for signal emission tower erection antenna or microwave transmission equipment, so this tower also called microwave communications tower

Production process: Communication tower, platform, by tower body lightning rods, ladder, antenna support etc steel components, and the hot DIP galvanized anti-corrosion processing,
Application: Suitable for various communication signal transmission project
Main technical parameters: Design height: 5 meters - 180 meters
Design wind speed: 25-80M/S
5mm 10mm wrapped ice: Seismic intensity: 8-60 appropriate temperature: -60 - 60
Vertical deflection < 1/1000: Grounding resistance≤ 4 Ω
Anticorrosive processing: Galvanized (normal corrosion life 30 years)

Single Port GSM Fixed Wireless Terminal (ETI603)

Model NO.:	ETI603
Productivity:	5, 000PCS/Month
Trademark:	OEM
Origin:	China
Usage:	Telephone
Type:	Router
Style:	Wireless
Export Markets:	North America, South America, Eastern Europe, Southeast Asia, Africa, Oceania, Mid East, Eastern Asia, Western Europe

Product Description

Features:
1. Based on GSM 900/1800/850/1900MHz
2. Support reverse polarity
3. Can connect to PBX.

Main Functions:
1. Suitable for GSM 900/1800/850/1900MHz, using SIM card
2. Making & answering calls after connecting to telephone set
3. Available to connect with telephone set, Billing machine, etc
4. Supports the functions of redialing and hands-free
5. Voice adjustment of calling in and out, multi-lever volume to be chosen
6. Caller ID
7. Featured with signal intensity indicator, working status indicator, breakdown indicator, etc
8. Built-in PIN code protection featured with locking device/card/network

Technique Parameters:
Working Frequency
Rx (EGSM 900): 925~960MHz
Tx (EGSM 900): 800~915MHz
Rx (DCS 1800): 1805~1880MH
Tx (DCS 1800): 1710~1785MHz
Power Consumption: <=5W
Working Voltage: DC5V
User Access Port:
Dialing tone frequency: 450Hz continued
Busy tone frequency: 450Hz 0.35s/0.35s

Package Information:
Size: 181(L) x 125(W) x 35 (H)mm
Weight: 275g
Package: 1PCS/box; 20boxes/carton
Carton size: 58.5*29.5*37.5CM
G. W/CTN: 15kgs

Channels VoIP GSM Gateway, VoIP ATA (GoIP4)

Product Description 4 Channels VoIP GSM Gateway (GoIP4), VoIP ATA
GoIP4 VoIP GSM Gateway bridges the GSM and the IP networks by enabling voice communications. It is ideal for VoIP to Local termination where a fixed telephone line (PSTN) is not available or for cellphone roaming via the a VoIP network. Significant savings on long distance charges can be realized.
Key Features

Open Standard VoIP Protocols (ITU H. 323 V4 and IETF SIP V2)
Single or Multiple Server Registrations
Two 10/100 Ethernet circuits connect to the LAN and an additional device
GSM module for making GSM calls
Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
VLAN and QoS support
NAT Transversal and Router functions
Voice prompts, HTTP Web, Auto Provision support for configuration and updates
Highly stable embedded Linux operating system in high performance ARM 9 Processor
Basic Features
LEDs for Power, Ready, Status, WAN, PC, GSM
Call forward from GSM to VoIP and VoIP to GSM
Dial in mode or dial out mode only
Dial Plan
Password protection for both GSM dial in or dial out
Retransmit GSM Caller ID to VoIP terminal
Enhanced Features
Dynamic selection of codec
Advanced jitter buffer
Automatic traversal of NAT and firewall
VLAN / Qos
Router
Echo cancellation for Speakerphone
Comfort noise generation (CNG)
Voice activity detection (VAD)
Auto provisioning (requires auto provisioning server)
On line firmware upgrade
Multi-language support: English and Chinese
Supported Standards
ITU: H. 323 V4, H. 225, H. 235, H. 245, H. 450
RFC 1889 - RTP/RTCP
RFC 2327 SDP
RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC 2976 SIP INFO Method
RFC 3261 SIP
RFC 3264 Offer/Answer model with SDP
RFC 3515 SIP REFER Method
RFC 3842 A Message Summary and Message Waiting Indicator
RFC 3489 Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)
RFC 3891 SIP Replaces Header
RFC 3892 SIP Referred-By Mechanism
Draft-ietf-sipping-CC-transfer-04 Session Initiation Protocol Call Control - Transfer
Codec: G. 711 (A/µ Law), G. 729A/B, G. 723.1
DTMF: RFC 2833, In-band DTMF, SIP INFO
Physical and Environmental

Operating temperature: 10° C to 40° C (50° F to 104° F)
Storage temperature: 0° C to 50° C (32° F to 122° F)
Weight: 500 g (1 lb) (Including AC/DC Adapter)
Size: 160 mm (W) x 252 mm (L) x 30 mm (H) (Width x Length x Heigth
Power: 12 Vdc 2A (AC/DC adapter included)
Safety:
Emissions:
Immunity:
Other Approval:
Warranty: One year

Processor	ARM9E133MHz
DSP	VPDSP101-4 100MHz
RAM	16M
FLASH	4M
Power	DC12V/2000mA +-10%	Input AC100V to AC240V
Consumption	Themaximum 5W
GSM Module Type	Default 900M/1800M Optional 850M/1900M
LED	RUN, GSM, LAN, PC	,
Network card	2 RJ45; Supported NAT	100/10BASE-T
Weight	900 grams	WHITOUT DC ADAPTER
Working temperature	0-42C
Working humidity	40%-90%Not congealed
Colour	Grey
GSMSIM PORT	4
VoIP Chunnel	4

4 Channels VoIP GSM Gateway, VoIP ATA (GoIP4)

Product Description 4 Channels VoIP GSM Gateway (GoIP4), VoIP ATA
GoIP4 VoIP GSM Gateway bridges the GSM and the IP networks by enabling voice communications. It is ideal for VoIP to Local termination where a fixed telephone line (PSTN) is not available or for cellphone roaming via the a VoIP network. Significant savings on long distance charges can be realized.
Key Features

Open Standard VoIP Protocols (ITU H. 323 V4 and IETF SIP V2)
Single or Multiple Server Registrations
Two 10/100 Ethernet circuits connect to the LAN and an additional device
GSM module for making GSM calls
Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
VLAN and QoS support
NAT Transversal and Router functions
Voice prompts, HTTP Web, Auto Provision support for configuration and updates
Highly stable embedded Linux operating system in high performance ARM 9 Processor
Basic Features
LEDs for Power, Ready, Status, WAN, PC, GSM
Call forward from GSM to VoIP and VoIP to GSM
Dial in mode or dial out mode only
Dial Plan
Password protection for both GSM dial in or dial out
Retransmit GSM Caller ID to VoIP terminal
Enhanced Features
Dynamic selection of codec
Advanced jitter buffer
Automatic traversal of NAT and firewall
VLAN / Qos
Router
Echo cancellation for Speakerphone
Comfort noise generation (CNG)
Voice activity detection (VAD)
Auto provisioning (requires auto provisioning server)
On line firmware upgrade
Multi-language support: English and Chinese
Supported Standards
ITU: H. 323 V4, H. 225, H. 235, H. 245, H. 450
RFC 1889 - RTP/RTCP
RFC 2327 SDP
RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC 2976 SIP INFO Method
RFC 3261 SIP
RFC 3264 Offer/Answer model with SDP
RFC 3515 SIP REFER Method
RFC 3842 A Message Summary and Message Waiting Indicator
RFC 3489 Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)
RFC 3891 SIP Replaces Header
RFC 3892 SIP Referred-By Mechanism
Draft-ietf-sipping-CC-transfer-04 Session Initiation Protocol Call Control - Transfer
Codec: G. 711 (A/µ Law), G. 729A/B, G. 723.1
DTMF: RFC 2833, In-band DTMF, SIP INFO
Physical and Environmental

Operating temperature: 10° C to 40° C (50° F to 104° F)
Storage temperature: 0° C to 50° C (32° F to 122° F)
Weight: 500 g (1 lb) (Including AC/DC Adapter)
Size: 160 mm (W) x 252 mm (L) x 30 mm (H) (Width x Length x Heigth
Power: 12 Vdc 2A (AC/DC adapter included)
Safety:
Emissions:
Immunity:
Other Approval:
Warranty: One year

Processor	ARM9E133MHz
DSP	VPDSP101-4 100MHz
RAM	16M
FLASH	4M
Power	DC12V/2000mA +-10%	Input AC100V to AC240V
Consumption	Themaximum 5W
GSM Module Type	Default 900M/1800M Optional 850M/1900M
LED	RUN, GSM, LAN, PC	,
Network card	2 RJ45; Supported NAT	100/10BASE-T
Weight	900 grams	WHITOUT DC ADAPTER
Working temperature	0-42C
Working humidity	40%-90%Not congealed
Colour	Grey
GSMSIM PORT	4
VoIP Chunnel	4

GSM Terminal With Pstn (GSM-ETI600)

Product Description Product Description
GSM terminal

1.CE compliant
2.GSM 900/1800MHz, 850/1900MHz and PSTN network
3.Making & answering calls after connecting to telephone set
4.Can make phone calls via GSM or PSTN network by choice or according to the routing parameters pre-defined on the route
5.Available to connect with telephone set, PBX, Billing machine, etc
6.Caller ID
7.Supports the functions of redialing and hands-free
8.Voice adjustment of calling in and out, multi-lever volume to be chosen
9.Featured with indicators of signal intensity, working status, breakdown, etc
10.Can proceed with setting up and modification of parameters through the phone set

Technique Parameters:
Working Frequency
Rx (EGSM 900):925~960MHz
Tx (EGSM 900):800~915MHz
Rx (DCS 1800):1805~1880MH
Tx (DCS 1800):1710~1785MHz
Power Consumption:<=5W
Working Voltage:DC12V
User Access Port:
Dialing tone frequency:450Hz continued
Busy tone frequency:450Hz 0.35s/0.35s

Package Information:
Volume:175(L) X 110 (W) X 30 (H) mm
Weight:629g
Package:1PC/box;20PCS/carton

8 SIM Cards VoIP GSM Gateway, VoIP Gateway GoIP8

Product Description 8 SIM Cards VoIP GSM Gateway, VoIP gateway GoIP8
GoIP8 VoIP GSM Gateway bridges the GSM and the IP networks by enabling voice communications. It is ideal for VoIP to Local termination where a fixed telephone line (PSTN) is not available or for cellphone roaming via the a VoIP network. Significant savings on long distance charges can be realized.
Key Features

VoIP GSM gateway, VoIP gateway

Open Standard VoIP Protocols (ITU H. 323 V4 and IETF SIP V2)
Single or Multiple Server Registrations
Two 10/100 Ethernet circuits connect to the LAN and an additional device
GSM module for making GSM calls
Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
VLAN and QoS support
NAT Transversal and Router functions
Voice prompts, HTTP Web, Auto Provision support for configuration and updates
Highly stable embedded Linux operating system in high performance ARM 9 Processor
Basic Features
LEDs for Power, Ready, Status, WAN, PC, GSM
Call forward from GSM to VoIP and VoIP to GSM
Dial in mode or dial out mode only
Dial Plan
Password protection for both GSM dial in or dial out
Retransmit GSM Caller ID to VoIP terminal
Enhanced Features
Dynamic selection of codec
Advanced jitter buffer
Automatic traversal of NAT and firewall
VLAN / Qos
Router
Echo cancellation for Speakerphone
Comfort noise generation (CNG)
Voice activity detection (VAD)
Auto provisioning (requires auto provisioning server)
On line firmware upgrade
Multi-language support: English and Chinese
Supported Standards
ITU: H. 323 V4, H. 225, H. 235, H. 245, H. 450
RFC 1889 - RTP/RTCP
RFC 2327 SDP
RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC 2976 SIP INFO Method
RFC 3261 SIP
RFC 3264 Offer/Answer model with SDP
RFC 3515 SIP REFER Method
RFC 3842 A Message Summary and Message Waiting Indicator
RFC 3489 Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)
RFC 3891 SIP Replaces Header
RFC 3892 SIP Referred-By Mechanism
Draft-ietf-sipping-CC-transfer-04 Session Initiation Protocol Call Control - Transfer
Codec: G. 711 (A/µ Law), G. 729A/B, G. 723.1
DTMF: RFC 2833, In-band DTMF, SIP INFO
Physical and Environmental

Operating temperature: 10° C to 40° C (50° F to 104° F)
Storage temperature: 0° C to 50° C (32° F to 122° F)
Weight: 500 g (1 lb) (Including AC/DC Adapter)
Size: 200 mm (W) x 350 mm (L) x 70mm (H) (Width x Length x Heigth)
Power: 12 Vdc 2A (AC/DC adapter included)
Safety:
Emissions:
Immunity:
Other Approval:
Warranty: One year

Processor	ARM9E133MHz
DSP	VPDSP101 95MHz	2 pieces
RAM	16M
FLASH	4M
Power	DC12V/3A +-10%	Input AC100V to AC240V
Consumption	Themaximum 20W
GSM Module Type	Default 900M/1800M Optional 850M/1900M
LED	RUN, GSM, LAN, PC	,
Network card	2 RJ45; Supported NAT	100/10BASE-T
Weight	1.204kg	WHITOUT DC ADAPTER
Working temperature	0-42C
Working humidity	40%-90%Not congealed
Colour	Gray
GSMSIM PORT	8
VoIP Chunnel	8

900/1800 GSM VoIP Gateway (WIA2001)

Product Description 1. SIP, MGCP protocols optional
2. 1 / 4 / 8 GSM/CDMA/WCDMA channel optional IMEI and PIN modified
3. SMS sending and receiving & SMS Opening API
4. Both central and standalone remote management
5. Proven integration with leading Softswitch and SIP Servers
6. High performance-price-ratio VoIP applications

Multiple models optional
WIA2000 series GSM/CDMA/WCDMA VoIP gateway has 3 available models: WIA2001, WIA2004, WIA2008. Also it has GSM/CDMA/WCDMA multi frequency bands customized.

One number access
You can setup one number as the access number and hide the other numbers, make it comfortable for VoIP business operation.

Service-orient Architecture, Rich Services Supported
It provide various services, for example, calling card, SMS. Service-orient architecture in line with the development evolution of telecom technology, satisfies the continuous expanding need in the future.

Easy Management & Maintenance
WIA2000 series VoIP gateway can be standalone managed, also can be central managed by SNMP/TR-069 based NMS.

Safety and Reliability
Administrator authentication mechanism; Intelligent multiple trunk routing technology, 8 years history, more than 100 thousand channels delivered.

SMS and SMS opening API supported
SMS sending and receiving, provides open API interface for third party to develop their own business.

IMEI & PIN Modified
WIA2000 series provide IMEI & PIN modified function, ensure GSM/CDMA gateway working well in different regions & environment.

Rich ethernet features, router and bridge modes optional
WIA2000 series gateway support router or bridge modes. It also supports rich features, for example, NAT, DHCP Server/Client, DNS, PPPoE client and so on.

High Performance-price-ratio
8 years mature hardware and software, large scale application, China R&D and manufacturing cost, make it cost-effective choice for customers all over the world.

	WIA2001	WIA2004	WIA2008
Interfaces
GSM/CDMA/WCDMA SIM Ports	1 port	4 ports	8 ports
VoIP channels	1 channel	4 channels	8 channels
Network interfaces	2*10/100M BASE-TX , RJ45	4*10/100M BASE-TX , RJ45	4*10/100M BASE-TX , RJ45
LED indicators	Power, Run, Signal strength indicator, LAN
Voice
Voice over packet Capabilities	G.168 with 32, 64 or 128 ms tail length
Voice Compression	G.711, G723.1, G.729 A/B
DHCP Client/Server	NAT Router or Switched Mode
Network Protocols	TCP/UDP, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, TELNET, PPPOE, STUN
QoS	Diff Serve, TOS, 802.1P/Q VLAN tagging
Signaling
DTMF Method	RFC 2833, SIP Info, INBAND
IP Signaling	SIP(RFC 3261) over UDP,MGCP
Provisioning	TFTP, HTTP
Security
Management	Web browser, Telnet, Voice prompt
Physical
Universal Power Supply	Output:12VDC Input:100-240VAC/50Hz	100-240VAC/50Hz
Environmental	Operational:32-104°For 0-40°C  Storage:14-140°For -10-60°C         Humidity:10%-90%
Dimensions
Weight
Additional Features
Caller ID	Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CID
Polarity Reversal	Yes
Homologation
Compliance	FCC, CE

RJ11 Phone Line Connector Surge Protector (DK-NDCT RJ11)

Product Description RJ11 Phone Line Connector Surge Protector

Applicable Scope
Phone, SPC exchange, Fax, Modem, Communication Terminal Equipment, Communication Room Protection

Character
1. Maximum Discharging Currents 10kA (8/20us)
2. High Speed Transmission Rate, Low Loss
3. Interference Absorption
4. Communication Line SPD
5. Various Working Voltage Available: DC24V, DC48V, DC200V
6. Easy Installation, Simple Maintenance

Brief Introduction
RJ11 Phone Line Connector Surge Protector is designed according to the IEC Communication SPD standards.

RJ11 Serials Products is suitable for the lightning protection of Phone Line, Dedicated Communication Line, SPC Exchange and any other equipments with RJ11 connector

The Connector is Standard Crystal RJ11 type. Single RJ11, 20 RJ11 ports and 33 RJ ports protection is available. Single port signal SPD can protect 1pair/2pars communication line. Multi ports product can be installed in the standard 19" cabinet, easy installing and maintenance.

8/20us Maximum Discharging Currents Imax can be 10kA/Line. Respond time can be Pico second level, and low residual Voltage.

Item	DK-nDCt/RJ11
Type	200 Signal SPD
Shell Material	Aluminum
Shell Protection Level	IP30
Working Termperture	-40 ºC —+85ºC
Relative Humidity	<95%
Maximum Working Voltage Uc (V)	220
Nominal Discharging Currents In(8/20us) (kA)	5
Maximum Discharging Currents Imax(8/20us) (kA)	10
Limiting Voltage(10/700us) (V)	≤300
Transmission Rate(bit/s)	2M
Inset Loss (dB)	0.5
Impedance (Omega)	10
Connector Type	RJ11
Weight (kg)	0.08
Dimension (mm)	59X25X25
Quantity of Core Line under Protection	3, 4
Art. No.	1008002

IP Phone (3 SIP line, IAX2, POE, VPN) (VI2008)

Product Description Business IP Phone


3 SIP Lines + IAX2 line
2 RJ 45 Ports & POE
Support VLAN, VPN(L2TP)
Large DOT matrix LCD display(57.54 x 79.2mm)
RJ9 Interface Headset

Advanced Application Features	Large dot matrix LCD display and softkeys make user easier to use
	2 headset interface ( 3.5mm Jack or RJ9 connector)
	6 memory keys
	Support POE
	Multilanguage realizes localization
	SIP application: support call forward/ transfer/ holding/ waiting/conference/
	Call control features: flexible dial map, hotline, empty calling no. reject service
	Incoming calls /outgoing calls / missing calls each support 100 records
	Support Phone Book 500 records
	Support conference and voice record on SIP server
	Support SMS
Network Features:	WAN/LAN: 10/100M Ethernet ports, supports Route and Switch mode.
	Support PPPoE for xDSL
	Support basic NAT and NAPT
	NAT transverse: support STUN client
	Support DHCP client on WAN
	Support DHCP server on LAN
	Support main DNS and secondary DNS server.
	Support VLAN
	Support DMZ
	Support DNS Relay, SNTP Client, Firewall
	Support VPN (L2TP)
	QoS with DiffServ
	Network tools in telnet server: including ping, trace route, telnet client
Maintenance and Management	Web, telnet and keypad management
	Management with different account right
	Upgrade firmware  through POST mode
	Telnet remote management/ upload/download setting file
	Safe mode provide reliability
	Support Auto Provisioning (upgrade firmware through HTTP, FTP or TFTP)
	Support Syslog
Basic Features	Support SIP 2.0 (RFC3261) and correlative RFCs
	Support IAX2
	SIP supports  3 SIP servers. Can connect to  3 SIP servers at the same time.
	Codec: G. 711A/u, G. 723, G. 729a/b, G. 722
	Echo cancellation: G. 168 Compliance in LEC, additional acoustic echo cancellation
	Support Voice Gain Setting, VAD, CNG
	Full duplex hands-free speakerphone
	SIP support SIP domain, SIP authentication (none, basic, MD5)
	DTMF MODE: support DTMF_RELAY, SIP_INFO, RFC_2833

Vonets SIP ATA (HT-600)

Product Description Product Characteristics

 (RFC3261), TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP/RARP, DNS, DHCP, NTP, STUN, and TFTP etc;
Use the DSP CNOS chip to keep a wonderful audio-quality, Jitter control and packet loss concealment technology;
Supports kinds of voice codes, including G. 711 (a-law and u-law) G. 723.1 (5.3K/6.3K) G. 726 (40K/32K/24K/16K), G. 729A/B, and iLBC;
Supports incoming call on show, restricting, and holding, disconnection, call transfer, call divert, DTMF, dialing plan, etc;
Supports conference call;
Supports NAT auto- penetrating, no need to modify the setting of the NAT;
Supports passing through and T38 fax, voice restrain and jerquers, CNG, echo restrains (G. 168), AGC;
Supports remote upgrade and upgrade configuration function;
Supports layer 2(802.1Q VLAN, 802.1p) and layer 3 (QoS, DiffServ, ToS);
Supports SIP encrypts mode, avoid force-out;
Supports upgrading encrypts configuration files by HTTP;
Supports configuration files by inside IVR equipment, Web browser, HTTP Center Server;
Supports the volume adjustment of gateway;

SIP 2.0	YES
Router/bridge	NO
DHCP	YES
EthernetPort	1WAN(RJ45)
FXS	1FXS
Size	60*110*25mm
Weight	0.33kg
Active status temperature	0-40°
Humidity	10-90%
Power  supply adapter	AC IN: 100V-240V  DC OUT: +9V/600mA Use power: < 2.5W
Authentication	FCC/CE

Fixed Wireless GSM 8-Channel VoIP Gateway (GOIP-8)

Produ

Packing:	Box Dimension: 39*22*9cm (L*W*H) G.W.:1.5kg
Model NO.:	GoIP-8
Standard:	Package: 10PCS/Carton
Productivity:	10, 000 PCS/Month
Unit Price:	USD 926.16/ Unit
Shipment Terms:	by Air
Trademark:	DBL
Origin:	China
Usage:	Telephone
Type:	VoIP Products
Style:	Wireless
Support:	Vlan and Qos,DHCP, Pppoe, Fix IP
VoIP Protocal:	H. 323&Sip
Build in:	Reply Encryption
Working Frequency:	Default 900m/1800m /Optional 850m/1900m
Warranty:	One Year
Export Markets:	North America, South America, Eastern Europe, Southeast Asia, Africa, Oceania, Mid East, Eastern Asia, Western Europe

ct Description The Fixed Wireless GSM 8-Channel VoIP Gateway (GOIP-8) bridges the GSM and the IP networks by enabling voice communications. It is ideal for VoIP to Local termination where a fixed telephone line (PSTN) is not available or for cellphone roaming via the a VoIP network. Significant savings on long distance charges can be realized.

Key Features
Open Standard VoIP Protocols (ITU H. 323 V4 and IETF SIP V2)
Single or Multiple Server Registrations
Two 10/100 Ethernet circuits connect to the LAN and an additional device
GSM module for making GSM calls
Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
VLAN and QoS support
NAT Transversal and Router functions
Voice prompts, HTTP Web, Auto Provision support for configuration and updates
Highly stable embedded Linux operating system in high performance ARM 9 Processor

Basic Features
LEDs for Power, Ready, Status, WAN, PC, GSM
Call forward from GSM to VoIP and VoIP to GSM
Dial in mode or dial out mode only
Dial Plan
Password protection for both GSM dial in or dial out
Retransmit GSM Caller ID to VoIP terminal

Enhanced Features
Dynamic selection of codec
Advanced jitter buffer
Automatic traversal of NAT and firewall
VLAN / Qos
Router
Echo cancellation for Speakerphone
Comfort noise generation (CNG)
Voice activity detection (VAD)
Auto provisioning (requires auto provisioning server)
On line firmware upgrade
Multi-language support: English and Chinese

Supported Standards
ITU: H. 323 V4, H. 225, H. 235, H. 245, H. 450
RFC 1889 - RTP/RTCP
RFC 2327 SDP
RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC 2976 SIP INFO Method
RFC 3261 SIP
RFC 3264 Offer/Answer model with SDP
RFC 3515 SIP REFER Method
RFC 3842 A Message Summary and Message Waiting Indicator
RFC 3489 Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)
RFC 3891 SIP Replaces Header
RFC 3892 SIP Referred-By Mechanism
Draft-ietf-sipping-CC-transfer-04 Session Initiation Protocol Call Control - Transfer
Codec: G. 711 (A/µ Law), G. 729A/B, G. 723.1
DTMF: RFC 2833, In-band DTMF, SIP INFO