Primatronix Nanho Technology Ltd.

This SIP phone combines the reliability of traditional telephones with the efficiency of internet-based technology, delivering a modern, flexible, and cost-effective solution for business communication. With minimal hardware and maintenance needs, it lowers long-distance and international call costs while enabling easy scaling by allowing businesses to add or remove users as needed. SIP Phone [Lower Communication Costs] Minimal hardware and maintenance requirements.[Scalable &amp; Efficient] Easily add or remove users as business needs evolve.[Wireless convenience] Perfect for spacious workplaces or settings where traditional wired connections are impractical.

SIP Phone

SIP Phone for modern, cost-effective business communication

Ningbo Leego Home Co., Ltd.

Terminal Type: Analog Phones, IP Phones, Video Phones, Soft Termi Single Set of User Capacity: 600~1, 000 (Max. 1,000 IP Users/Simulated Users) Trunk Interfaces: Maximum 36CH FXO, 4e1/T1, 4bri Signaling Protocol: Pri,Qsig,Ss7,R2,SIP,Bri Specification: 86.1mm*447mm*708mm  

Voice Gateway

Enlighten Company Limited

Model: EL-WCP-IPInput power: PoE(802.3af)Protocol: TCP-IPVoIP protocol: SIPEnvironment Noise CancellationEcho CancellationIP-65 water-proofRed re-assurance visible LEDBig alarm button with Reset

IP based Call Point with VoIP

Shenzhen Chengfenghao Electronics Co., Ltd

 	 Support Frequency Band TDD 38/39/40/41 	 FDD 1/3/5/7/8 	 WCDMA 1/2/5/8 	 GSM 2/3/5/8 	 Language&nbsp; Portuguese, English, Spanish, Russian and so on 	 Call log 20pcs (dialed/ received/ missed). 	 Supports VOLTE (HD Voice Calling service on 4G LTE) 	 Supports WiFiLi-Ion battery 3.7V 1800mA  

4G Fixed Wireless handset phone

4G Fixed Wireless handset phone with VOLTE and HD Voice

Shenzhen Olycom Technology Co., Ltd.

Video IP Phone 

Auraland Electronics Co Ltd

Feature is a Internet Telephone.   3 VolP accounts, hotline   Call waiting, call transfer, call forward, Hold, mute, flash, auto-answer, redial, 3-way conference, DND, speed dial   Phonebook with 140 groups memory   Volumn adjustment, ring tone selection, tone scheme, System log, DDNS function  

SIP/VoIP Internet Telephone

SIP VoIP Internet Telephone with multiple features

Suntra Int'l Development Ltd

  SIP standardization deployed throughout  Sustain VoIP calls in noisy unmanaged networks, supporting up to 40% packet loss and up to 500ms of latency  Supports TOS bit for improved QoS in managed networks (Differentiated services of "DiffServ")  Voice packetization & flexible compression  Dynamic & automatic negotiation of codec selection  LAN bridge   built in 2 port ethernet hub   (Optional)  Caller ID receive and display  Alphanumeric LCD  Hands free dialing  Phone book memory dialing  Incoming, outgoing and missed call logs  

VoIP SIP Phone

VoIP SIP Phone for high-quality internet telephony

Shenzhen Voptel Techonology Co Ltd

   VI2006 is IP phone ideal for service provider and small offices. Enterprise SIP phone's new design and enhanced features address the need for an elegant IP handset solution for the executive office at a highly competitive price.   VI2006 is an Internet based voice network phone terminal, which adopts multiple voice control protocols and voice compression coding methods to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service.   VI2006 supports IA x 2 &amp; SIP protocols, support bridge and router model. Support IA x 2 and dual public server, offers two Ethernet interfaces and is compatible with various softswitch systems and VoIP voice gateways to provide broadband IP voice service.   VI2006 supports traditional telephony features such as flexible dial map, hotline, empty calling reject, black list for reject authenticated call, limit call, no disturb, caller ID as well as advanced enterprise features such as 2 independent SIP accounts, 3-way and conferencing.  Specifications:
   Support SIP 2.0 (RFC3261) and correlative RFCs.   Codec: G.711 A/U law, G.723.1, G.729a/b, G.722, G.722.1, G.726.   Echo cancellation: support G.168, and hands-free can support 96ms, hand free speaker phone.   Support voice gain setting, VAD, CNG.   Full duplex hands-free speaker phone.   NAT transverse: support STUN client.   SIP support SIP domain, SIP authentication (none, basic, MD5), peer to peer/IP call.   SIP support 2 SIP lines. Can connect to SIP1 and SIP2 server at the same time.   DTMF: support SIP info, DTMF relay, RFC2833.   SIP application: support call forward/transfer/holding/waiting/3 way talking.   Call control features: flexible dial map, hotline, empty calling reject, black list for reject authenticated call, limit call, no disturb, caller ID.   Support phonebook 500 records incoming calls/outgoing calls/missing calls.   Each support 100 records.   Support 10 kind of ring type and 1 user ring.   Support SRTP.   Support SMS.   Support speed dial.  Network features:
   Support basic NAT and NAPT.   Support PPPoE for xDSL.   Support DHCP client on WAN.   Support DHCP server on LAN.   Support VLAN (optional: voice VLAN/data VLAN) QoS with DiffServ.   Support DMZ.   Support main DNS and secondary DNS server.   Support DNS relay, SNTP Client, Firewall.   Network tools in telnet server: Including ping, trace route, telnet client.  Maintenance &amp; management:
   Web, telnet and keypad management.   Management with different account right.   Upgrade firmware through HTTP, FTP or TFTP. Telnet.   Remote management/upload/download setting file.   Safe mode provide reliability.   Support auto provisioning (upgrade firmware or configuration file).   Support syslog.  

Low Cost VoIP Phone in Promotio

Low Cost VoIP Phone - VI2006 for small offices

Huizhou City Huiyang District Meisiqi Industry Development Co., Ltd.

MSQ VoIP phone series are feature-rich, cost-effective SIP phones specially designed for ITSPs, small to medium businesses and residential application. It supports 5 user accounts, auto-provision as well as zero configuration and easy maintenance. It offers high quality audio, a broad range of voice codes and phone features to general level users. 
   Two RJ45 ports with router built-in, one for Internet, one for PC.   One RJ11 port FXO port.   POE (power over Ethernet).  Network protocol: 
   SIP V1(RFC2543), V2(RFC3261).   IP/TCP/UDP/RTP/RTCP.   IP/ICMP/ARP/RARP/SNTP.   TFTP client/DHCP client/PPPoE client.   Telnet/HTTP server.   DNS client.   NAT.  SIP server:
   Registrar server: 5 SIP user accounts.   Outbound proxy mode.   Power over Ethernet (optional).  Code:
   G.711.   G.723.1/iLBC.   G.726.   G.729A.   G.729B adds VAD, CNG to G.729.  Configuration: 
   Web browser.   Console/telnet.   IVR/keypad.   Auto-provision.  IP assignment: 
   Static IP.   DHCP.   PPPoE.  NAT traversal: 
   STUN.   QoS.   ToS field.  Dial method: 
   Direct IP call without SIP proxy.   Dial number via SIP server.   Dial URL via SIP server.  DTMF function: 
   In-band DTMF.   Out-of-band DTMF.   SIP info.  Phone features: 
   Caller ID.   Call hold.   Call waiting.   Call forward.   Speaker phone.   3-way conference.   Redial and flash.   Speed dial.   Phone book.   Volume adjustment.   Ring tone selection.  Voice quality: 
   VAD.   CNG.   LEC.   Packet loss compensation.   Adaptive jitter buffer.  Firmware upgrade: 
   TFTP.   Console.   HTTP.   Auto-provision.  Security: 
   HTTP 1.1 basic/digest authentication for web setup.   MD5 for SIP authentication (RFC 2069/RFC 2617).  Interface: 
   Power input: 100-240V AC, output power: DC-5.5V, 1A.   WAN RJ45, 10/100M port.   LAN/PC RJ45, 10/100M port.   TEL IN (FXO) RJ11 port.  Certificate:
   CE.   FCC.  

VoIP SIP Phone POE with PSTN 

HK Retail Technology Industry Association Ltd/GS1 HK

The wafer VoizMate is a daily Cisco IP phone application platform. It combines with the Cisco IP telephony solutions to make full use of IP phone communications for enterprises including more value-added services. It can satisfy more requirements of corporation for service integration and information bulletin, finally improving the corporate efficiency and competitiveness.
 
  Combines with the Cisco IP telephony solutions to make full use of IP phone communications for enterprises. 
  High quality. 
 

Wafer VoizMate

Wafer VoizMate is a Cisco IP phone application platform.

Suzhou International Science Park

This outlined IP video phone is designed with advanced features. The large LCD touch screen and keypads ensures enhanced user friendliness. The phonebook can save up to 2,000 records. This IP video phone is ideal for office use.
 
  Telephone features: forward and transfer calls, conference call, paging and intercom, etc. 
  Blacklist, whitelist, DND. 
  Call history: incoming, outgoing and missed (each 1,000 entries). 
  Import and export phonebook. 
  External phone book: XML. 
  Photo frame. 
 

IP Video Phone

IP Video Phone with advanced features for office use

   VI2007 is SIP phone ideal for both the small office and general enterprise users. Enterprise SIP phone's new design and enhanced features address the need for an elegant IP handset solution for the executive office at a highly competitive price.   VI2007 is an Internet based voice network phone terminal, which adopts multiple voice control protocols and voice compression coding methods to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service.   VI2007 supports IAX2 &amp; SIP protocols, support bridge and router model. Support IA x 2 and dual public server, offers two Ethernet interfaces and is compatible with various soft switch systems and VoIP voice gateways to provide broadband IP voice service.   VI2007 supports traditional telephony features such as flexible dial map, hotline, empty calling reject, black list for reject authenticated call, limit call, no disturb, caller ID as well as advanced enterprise features such as 2 independent SIP accounts, 3-way and conferencing.  Specifications:
   Support SIP 2.0 (RFC3261) and correlative RFCs.   Codec: G.711 A/U Law, G.723.1, G.729a/b, G.722, G.722.1, G.726.   Echo cancellation: support G.168, and hands-free can support 96m, hand free speaker phone.   Support voice gain setting, VAD, CNG.   Full duplex hands-free speaker phone.   NAT transverse: support STUN client.   SIP support SIP domain, SIP authentication (none, basic, MD5), Peer to Peer/IP call.   SIP support 2 SIP lines. Can connect to SIP1 and SIP2 server at the same time.   DTMF: support SIP info, DTMF Relay, RFC2833.   SIP application: support call forward/transfer/holding/waiting/3 way talking.   Call control features: flexible dial map, hotline, empty calling reject, black list for reject authenticated call, limit call, no disturb, caller ID.   Support phonebook 500 records incoming calls/outgoing calls/missing calls.   Each support 100 records.   Support 10 kind of ring type and 1 user ring.   Support SRTP.   Support SMS.   Support speed dial.  Network features:
   Support basic NAT and NAPT.   Support PPPoE for xDSL.   Support DHCP client on WAN.   Support DHCP server on LAN.   Support VLAN (optional: voice VLAN/data VLAN) QoS with DiffServ.   Support DMZ.   Support main DNS and secondary DNS server.   Support DNS relay, SNTP client, firewall.   Network tools in telnet server: including ping, trace route, telnet client.  Maintenance &amp; management:
   Web, telnet and keypad management.   Management with different account right.   Upgrade firmware through HTTP, FTP or TFTP. Telnet.   Remote management/upload/download setting file.   Safe mode provide reliability.   Support auto provisioning (upgrade firmware or configuration file).   Support syslog.  

SIP Phone VI2007 for enterprise and small office use

MSQ VoIP gateway series are feature-rich, cost-effective SIP gatway specially designed for ITSPs, small to medium businesses and residential application. It supports 5 user accounts, auto-provision as well as zero configuration and easy maintenance. It offers high quality audio, a broad range of voice codes and phone features to general level users. 
   Two RJ45 ports with router built-in, one for Internet, one for PC.   2 FXS + 1FXO port.  Network protocol: 
   SIP V1(RFC2543), V2(RFC3261).   IP/TCP/UDP/RTP/RTCP.   IP/ICMP/ARP/RARP/SNTP.   TFTP client/DHCP client/PPPoE client.   Telnet/HTTP server.   DNS client.   NAT.  SIP server:
   Registrar server: 5 SIP user accounts.   Outbound proxy mode.  Code:
   G.711.   G.723.1/iLBC.   G.726.   G.729A.   G.729B adds VAD, CNG to G.729.  Configuration: 
   Web browser.   Console/telnet.   IVR/keypad.   Auto-provision.  IP assignment: 
   Static IP.   DHCP.   PPPoE.  NAT traversal: 
   STUN.   QoS.   ToS field.  Dial method: 
   Direct IP call without SIP proxy.   Dial number via SIP server.   Dial URL via SIP server.  DTMF function: 
   In-band DTMF.   Out-of-band DTMF.   SIP info.  Voice quality: 
   VAD.   CNG.   LEC.   Packet loss compensation.   Adaptive jitter buffer.  Firmware upgrade: 
   TFTP.   Console.   HTTP.   Auto-provision.  Security: 
   HTTP 1.1 basic/digest authentication for web setup.   MD5 for SIP authentication (RFC 2069/RFC 2617).  Interface: 
   Power input: 100-240V AC, output power: DC-12V, 1A.   WAN RJ45, 10/100M port.   LAN/PC RJ45, 10/100M port.  Certificate:
   CE.   FCC.  

VoIP Gateway SIP IP ATA 

VoIP Gateway SIP IP ATA for ITSPs and businesses

This Internet telephone adaptor comes with a network interface set up by network for DHCP, PPPoE, or fixed IP network environment for connection to the network. It comes with auto configuration support (plug-n-play for no user setup when network access is available). Smart firmware management support is provided to&nbsp;eliminate the need for manual firmware upgrade.   Message prompts for ATA status (report IP address).   Compatibility in multi- vendor open architecture VoIP networks.   Dynamic DNS (DDNS) support for the use of domain name instead of IP address.   NTP and redundant DNS servers support .   QOS support (Diff-Serve, 802.1p/q VLAN).   NAT/ firewall traversal support with session border controller or stun server.   SIP standardization deployed throughout.   Advanced audio processing - jitter buffer with packet delay and packet loss concealment.   Voice packetization and flexible codec compression.   Dynamic and automatic negotiation of codec selection.   LAN Bridge &ndash; built in 2-port Ethernet hub.   FXS port for interface to a normal PSTN phone.   Bypass port for interface to a PSTN line (Optional).   PSTN caller ID signal generation for VoIP calls.   Calls from FXS port to both VoIP and PSTN networks.   Call transfer, call forward, call waiting for VoIP calls.   Easy local user customization of options - via web browser interface.   Open system SIP VoIP architecture for seamless integration into multi-vendor VoIP networks.   Standards include: SIP 2.0, TCP/UDP/IP, RTP/ RTCP, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, MD5 authentication.   Codec supported: G.711(a-law/u-law), G.729A/B, G.723.1, GSM FR   DTMF dialing in the format of RFC 2833, SIP Info, or In-band DTMF signals.  

Analog Telephone Adaptor

Analog Telephone Adaptor for VoIP and PSTN integration

 This high-performance SIP phone is widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol. It features&nbsp;multiple lines that register to a single server on a multiple-server mode. It is equipped with an extra-large, easy-to-read LCD display and a headset jack. 
 
  With SIP standardization deployed throughout. 
  Optional feature: PoE. 
 

SIP Telephone

SIP Telephone for multimedia communication over Internet Protocol

CEM Solutions Pvt Ltd.

   Stand Alone, powerful voice gateway supports up to 50 simultaneous calls on PSTN/IP link and 5 VoIP registration &amp; VoIP Trunks. No additional licenses required for codec transcoding or web interface   Our Scalable VoIP PRI Gateways ranging from 1 to 2 PRI ports developed to enable direct connection of VoIP traffic to the PSTN/fixed line ISDN for extending communications and reducing call charges  Key Features 
   E1/T1 PRI Gateway; up to 2 SPAN interface   Built In Firewall   Flexible Call Routing   Transcoding (Voice coding G.711 A-law/u-law, G.722 G.729AB) licenses included.   Fax Pass-through.   Stand Alone with Web User Interface.   128 ms DSP based Echo Cancellation  Additional Features 
   Up to 50 simultaneous VoIP to ISDN PRI calls   2 T1/E1/J1 PRI ports with programmable NET/CPE modes   Support SIP v2 (RFC 3261 complaint)   Register with multiple SIP service providers   ISDN network clock synchronization for error-free communication   Interoperable with all the major-brand soft switches and IP-PBXs   PIN authentication for accessing Trunks to prevent unauthorized usage   Call detail records   Web-based configuration and management  

ISDN PRI VoIP Gateway

ISDN PRI VoIP Gateway supports up to 50 simultaneous calls

Shenzhen Baoan Gaoke Electronics Co Ltd

Pictured is a VoIP phone with a hands-free dialing function. This VoIP phone will&nbsp;calculate conversation time automatically. The trendy design easily matches any modern office decor.   Supports&nbsp;DNS call waiting function.   Protocol support: SIP (RFD2543, ARFC3261).   Three-way conversation.  

VoIP Phone

VoIP Phone with hands free dialing and modern design

The pictured IP video phone is fitted with the professional Broadcom VoIP and video chipsets chipsets. It is equipped with a 7&quot; digital 800 x 480 TFT LCD touch screen for brilliant performance. The IP video phone is an ideal choice for business meeting use.
 
  Android OS. 
  SIP supports 5 SIP servers. 
  VoIP/PSTN dual modules supported. 
  Visual angle adjustable display screen. 
  Rotatable CMOS sensor camera with 2 mega pixels. 
  Video resolution: QCIF, CIF, 4CIF, DCIF. 
  Image format: JPEG, PNG, GIF, BMP. 
  Support phonebook 2,000 records. 
 

IP Video Phone with VoIP and video capabilities

VI2006 is ideal for both the small office and general enterprise users.Enterprise SIP phone's new design and enhanced features address the need for an elegant IP handset solution for the executive office at a highly competitive price.


VI2006 is an Internet based voice network phone terminal, which adopts multiple voice control protocols and voice compression coding methods to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service.


VI2006 supports IAX2&amp;SIP protocols,support Bridge and Router model.Support IAX2 and dual public server, offers Two Ethernet interfaces and is compatible with various softswitch systems and VoIP voice gateways to provide broadband IP voice service.


VI2006 supports traditional telephony features such as Flexible dial map, Hotline, Empty calling reject, Black list for reject authenticated call, limit call, No disturb, Caller ID as well as advanced enterprise features such as 2 independent SIP accounts, 3-way and conferencing. 

Specifications

Support SIP 2.0 (RFC3261) and correlative RFCs
Codec:G.711 A/U Law, G.723.1, G.729a/b,G.722,G.722.1, G.726
Echo cancellation: Support G.168, and Hands-free can support 96ms, Hand free Speaker Phone
Support Voice Gain Setting, VAD, CNG
Full duplex hands-free speakerphone
NAT transverse:support STUN client
SIP support SIP domain, SIP authentication (none,basic, MD5), Peer to Peer/ IP call
SIP support 2 SIP lines. Can connect to SIP1 and SIP2server at the same time
DTMF:Support SIP info, DTMF Relay, RFC2833
SIP application: support Call forward/ transfer/holding/ waiting / 3 way talking
Call control features: Flexible dial map, Hotline,Empty calling reject, Black list for reject authenticatedcall, limit call, No disturb, Caller ID
Support Phonebook 500 recordsIncoming calls / Outgoing calls /Missing calls. 
Each support 100 records
Support 10 kind of ring type and 1 user ring
Support SRTPSupport SMS
Support Speed Dial

Network Features

Support basic NAT and NAPT
Support PPPoE for xDSL
Support DHCP Client on WAN
Support DHCP Server on LAN
Support VLAN (optional: voice vlan/data vlan) QoS with Diff Serv
Support DMZSupport main DNS and secondary DNS server
Support DNS Relay, SNTP Client, Firewall
Network tools in telnet server: Including ping, trace route,telnet client

Maintenance&amp;Management

Web ,telnet and keypad management
Management with different account right
Upgrade firmware through HTTP, FTP or TFTP. Telnet 
Remote management/Upload/download setting file
Safe mode provide reliability
Supoort Auto Provisioning (upgrade firmware orconfiguration file)
Support Syslog

 

IP Phone

VI2006 IP Phone with advanced features for enterprise users

Allodotcom Inc.

  - As shown is a telephone adapter.  - The telephone adapter is equipped with a 2x ethernet LAN interface - RJ45.  - Out-of-band DTMF (RFC 2833)   - 2 FXS ports & 1 FXO port  - Adaptive jitter buffer  - SIP Phone Number per FXS port  - User-friendly web panel  - DTMF out-of-band (RFC 2833)  - DHCP server for LAN  - Operating System: Linux 2.4.26  - IP Address Reservation by MAC Address  - Simultaneous active calls on FXS1 and FXS2  - PSTN pass-through  - Firmware upgrade via RFC 1867 upload  - SIP compatible  

VoIP Analog Telephone Adapter

VoIP Analog Telephone Adapter with multiple features

Fanvil Technology Co., Ltd.

 The IP phone is a sophisticated business VoIP communication terminal. The large dot matrix&nbsp;LCD display and soft-keys make&nbsp;it easy to use. This versatile VoIP phone is perfect for executives, office managers or administrative assistants who make and receive a large amount of calls. 
 
  2 headset interfaces (3.5mm jack or RJ9 connector). 
  6 memory keys. 
  Supports PoE. 
  Multi-language realizes localization. 
  SIP&nbsp;application: supports call forward/transfer/holding/waiting/conference, etc. 
  Flexible dial map and&nbsp;hotline. 
  Empty calling, black list and&nbsp;limit call. 
  No disturb function and caller ID display. 
  Incoming calls/outgoing calls/missing calls, each supports 100 records. 
  Supports phonebook of 500 records. 
  Supports conference and voice record on SIP server. 
 

IP Phone for sophisticated business VoIP communication

Neo Telecom Corporation

This wireless phone enables the high-quality VoIP service through a wireless-G network. It boasts unique features, such as 500 phone book memories and&nbsp;50 call histories. This functional cordless phone is perfect for busy families or offices.
 
  Call time display. 
  Dialed number display. 
  Redial function. 
  Call transfer, holding, waiting and forwarding functions. 
  SMS. 
  Language: English and Korean. 
  1.8&quot; color TFT LCD screen. 
  Resolution: 128 x 160. 
 

Wireless VoIP Phone

Wireless VoIP Phone for high quality communication

United Network Solutions Ltd

  MultiTech Hybrid VoIP Gateways (8 Ports).  Support DID/DDI, FXS, FXO, E&amp;M, T1 and E1 interfaces.  Programmable DID/FXS/FXO/E&amp;M Port.  

MultiVOIP(R)

MultiVOIP R Hybrid VoIP Gateway with 8 Ports

Vogtec (H.K.) Co., Limited

VoIP Desktop T12 	 Generous 128x64 pixel back light graphical LCD display or 2.8&#39; TFT color LCD 	 Multi Languages 	 SIP (RFC3261),TCP/IP/UDP, RTP, HTTP, ARP, ICMP, DNS (A record and SRV),DHCP(both client and server), PPPoE, TFTP, NTP 	 NAT traversal (STUN, etc), server fail-over, SIP presence (SIMPLE), and more 	 2 lines indicators with individual SIP account profiles 	 Soft keys, navigation keys and Line Keys can defined by user 	 Call waiting, hold, mute, transfer (blind or attended), forward, 3-way conferencing and more 	 Full duplex speaker phone w/advanced audio echo cancellation 	 Headset jack RJ9 for broad headset compatibility 	 Large phonebook (up to 500 contacts) and call history (up to 100 records) 	 Automated provisioning using TR-069 or encrypted XML configuration file, SRTP and TLS for advanced security protection, 802.1x for media access control 	 And many more enterprise grade features  

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