电子产品

搜罗全球顶尖电子产品公司提供的电子产品资讯。您可在贸发网采购平台即时谘询可靠和专业的产品供应商，以获取详尽的资料及报价。

电子产品供应商和制造商

Electronics

电讯服务

搜罗全球顶尖电讯服务公司提供的电讯服务资讯。您可在贸发网采购平台即时谘询可靠和专业的产品供应商，以获取详尽的资料及报价。

电讯服务供应商和制造商

Telecommunication

电话

elec

搜罗全球顶尖电话公司提供的电话资讯。您可在贸发网采购平台即时谘询可靠和专业的产品供应商，以获取详尽的资料及报价。

电话供应商和制造商

Phones

风青实业股份有限公司

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

深圳市诚丰浩电子有限公司

10.1' Tablet Phone Android 10.0 PSTN+ 4G+SIP With handset (RJ9) 4G RAM+32G ROM (8G+64G optional) HD quality camera Support WiFi 2.4G /5G Desk / Wall mount Support GMS Support Google play store  

Smart Android Tablet Phone SIP Phone support 4G network

10.1' Tablet Phone Android 10.0 Bluetooth cordless handset PSTN+ 4G 4G RAM+32G ROM (8G +64G optional) HD quality camera Support WiFi 2.4G /5G Support BT 5.0 Desk / Wall mount Support GMS Support Google play store 

Smart Android Tablet Telephone with Handset and Video call

太阳国际发展有限公司

  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

深圳市沃宇科技有限公司

   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.  

One Source Telecom Limited

Featured here is an SIP phone that enables enterprises to take full advantage of the benefits of the VoIP telephony. This SIP phone&nbsp;has all capabilities that are necessary for daily business activities, such as call hold, call transfer, call swap, call list and 3-party conferencing functions.
 
  Telephone book with up to 100 entries. 
  Hand-free speaking, connection for microphone and headphone. 
  Function keys and programmable speed dials. 
 

Eware Networks Limited

The SIP-T46G is Yealink latest revolutionary IP Phone for executive users and busy professionals. New designs appears the commerce, with highresolution TFT color display, delivering a rich visual experience. Yealink Optima HD technology enables rich, clear, life-like voice communications. Supports Gigabit Ethernet, a variety of device connections, including EHS headset and USB. With programmable keys, the IP Phone supports vast productivity-enhancing features.


 Features  	 Revolutionarily new design 	 Yealink Optima HD voice 	 Dual-port Gigabit Ethernet 	 4.3&quot; 480 x 272-pixel color display with backlight 	 Built-in a USB port, support Bluetooth headset (Through USB Dongle) 	 Up to 16 SIP accounts 	 Paper label free design 	 PoE(802.3af) class support 	 Headset, EHS support 	 Supports expansion modules 	 Stand with 2 adjustable angles 	 Wall mountable 	 Simple, flexible and secure provisioning options  
 Specification 

Audio Features 	 HD voice: HD handset, HD speaker 	 Codecs: G.722, G.711(A/&mu;), G.723, G.729AB, G.726, iLBC 	 DTMF: In-band, Out-of-band(RFC 2833) and SIP INFO 	 Full-duplex hands-free speakerphone with AEC 	 VAD, CNG, AEC, PLC, AJB, AGC  
Phone Features 	 16 VoIP accounts 	 Call hold, mute, DND 	 One-touch speed dial, hotline 	 Call forward, call waiting, call transfer 	 Group listening, SMS, emergency call 	 Redial, call return, auto answer 	 3-way conferencing 	 Direct IP call without SIP proxy 	 Ring tone selection/import/delete 	 Set date time manually or automatically 	 Dial plan 	 XML Browser 	 Action URL/URI 	 RTCP-XR (RFC3611), VQ-RTCPXR (RFC6035)  

Yealink SIP-T46G Ultra-Elegant Gigabit IP Phone

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 

 	 Features Yealink HD Voice 	 Wall mountable  

Yealink SIP-T23P Enterprise HD IP Phone


   This is a business class (SIP) IP telephone.  The phone features handsfree dialing.  Alphanumeric LCD  Lan bridge build in 2 port ethernet hub  Phonebook memory dialing  SIP sandardization deployed throughout  
 

Business Class (SIP) IP Telephone

 	 Features Yealink Optima HD voice 	 Integrated stand with 2 adjustable angles 	 Wall mountable  

Yealink SIP-T27P Enterprise HD IP Phone

 	 Features Yealink Optima HD voice 	 Stand with 2 adjustable angles 	 Wall mountable  

Yealink SIP-T29G Enterprise HD IP Phone

 	 Features revolutionarily new design 	 Supports expansion modules 	 Wall mountable  

Yealink SIP-T48G Ultra-Elegant Gigabit IP Phone

Kingtel Corporation

   The phone uses SIP protocol.   The IP phone complies with all well-known IP-PBX brands.   The phone supports auto-provision.   The phone is rated as high-quality speakerphone as commercial grade.   The phone supports POE-power over ethernet.   Single line speaker phone.  

IP Phone

 	 Features Full-HD 1080p30 video call Five-way audio/video mixed conferencing  

Yealink SIP VP-T49G Revolutionary Video Collaboration Phone

 	 Features revolutionarily new design 	 Yealink Optima HD voice integrated 	 Stand with 2 adjustable angles 	 Wall mountable  

Yealink SIP-T42G Ultra-Elegant Gigabit IP Phone

Suzhou International Science Park

This IP phone can register up to 4 SIP accounts. It features a large LCD screen and functional keys for added performance. This IP phone is convenient to use.
 
  SIP 2.0 (RFC3261). 
  HD voice on all lines. 
  Echo cancellation: G.168 compatible with the LEC, including hands-free (AEC). 
  VGS, VAD, CNG. 
  Speakerphone full-duplex. 
  Phonebook up to 500 entries. 
  Customized logos are welcome. 
 

   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.   Due toits excellent cost-performance ratio and high performance, it is absolutely a good choice for you.   Two RJ45 ports with router built-in, one for internet, one for PC.   POE (power over Ethernet) optional.  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.   Certificate: CE, FCC  

IP Phone with POE and PSTN Port

VTECH HOLDINGS LTD

The Group&#39;s diverse collection of telecommunication products elevates both home and business users&#39; experience and sells under the leading brand names VTech, Snom and AT&amp;T.
&nbsp;&nbsp;
 1. Residential Phones 
that are a perfect blend of design and functionality.
&nbsp;
 2. Commercial Phones 
that are easy to install and set up, being specially designed for businesses from small-to-medium sized operations to enterprise-level corporations. 	 Conference Phone 	 SIP Phone 	 Small-to-Medium Business Phone 	 SIP-DECT Cordless Handset 	 Hotel Phone 	 Cordless Headset  &nbsp;
 3. Other Telecommunication Products 
that cover a wide range of products which cater to various needs of home users, making life more comfortable. 	 Wireless Monitoring System 	 Safe&amp;Sound Baby Monitor 	 CAT-iq 2.0 Certified Handset 	 Integrated Access Device 	 CareLine 	 Baby Soother  

Telecommunication Product

深圳市宝安高科电子有限公司

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

   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

Asterisk, Elastix, 3CX compatible.
VoIP phone with 3 SIP lines &amp; 1 IAX2 line.
Support VLAN, VPN (L2TP), POE.
RJ9 headset.
2RJ 45 ports.
Large dot matrix LCD display.
 
Basic features: 
   Support SIP 2.0 (RFC3261) and correlative RFCs.   Support IA x 2.   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 speaker phone.   SIP support SIP domain, SIP authentication (none, basic, MD5).   DTMF mode: support DTMF_RELAY, SIP_INFO, RFC_2833.   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.   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.   Multi language 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.   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.  

VoIP Phone Compatible with Asterisk

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

 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. 
 

我的采购