Performance and features:
* SIP v1 (RFC2543), v2(RFC3261)
* Support Route,Two 10/100Mbps MACs
* Support T.38(Doing)
* IP/TCP/UDP/RTP/RTCP
* IP/ICMP/ARP/RARP/SNTP
* TFTP Client/DHCP Client/ PPPoE Client
* Telnet/HTTP Server
* DNS Client
* NAT/DHCP Server
* Support G.723.1 5.3k/6.3k,G.729,G.711 A-Law,¦Ì-Law audio codec algorithm
* Dynamic voice detection Echo cancellation Comfort noise generation
* Tone generation and Local DTMF generation and detection according with ITU-T
* Settings by HTTP web browser (IE6.0)
* Advanced settings by Telnet
* Voice prompt
* Upgrade by TFTP
* 2RJ45 Ports,Built-in Router,conference.Auto-provision or updating by HTTP,FTP or TFTP.
* For each GT-P302,it can have 5-SIP account and one PSTN phone number,that means each phone own 6 phone numbers,all can be used as callee at anytime.Fo the caller,these 6 account can be selected by dial different relevant prefix(including switch to PSTN as an ordinary PSTN phone)
* GT-P302 has a real FXO port to support router call from PSTN to VOIP or VOIP to PSTN.
Main technical index:
* Main chip: 32-bit RISC CPU with 125MHz clock rate
* Data storage: 2MB SDRAM
* Program memory: 1MB Flash memory
* Application Network environment: Two 10/100Mbps Fast Ethernet MAC
* Echo cancellation: G165 16ms
* Store quick dial number: 100
* Record phone number of missed call: 80
* Power loss: 2.7W(max)
* Power adapter: input AC 220V,output DC 9V 500mA
* Employing condition:
* Ambience temperature 0-40C(32-104F)
* Relative humidity 10-95%
* Atmosphere pressure 86-106Kpa
* Overall size: 220, 170, 70mm(L, W, H).
Solution: broadcom,
Support sip protocols
Support 2 sip lines thru. 2 different
Support voice mail, call
Waiting/transfer/holding/conference.
Support voice gain setting, jitter buffer, vad and cng.
Web configuration through built-in web server.
Upgrading firmware and configurations by http, ftp or tftp.
Dhcp client on wan and dhcp server on lan.
1. Standalone operation, no computer needed;
2. Uses broadband cable/xdsl or network connection for high quality voice over ip;
3. Supports sip, h.323 v4, mgcp, iax2, net2phone etc protocol;
4. Large lcd screen display;
5. With single/double rj45 broadband socket;
6. With h.323 proxy server inside;
7. Startup h.245 tunnel quickly;
8. Supports h.245 and q.931 keypad out-band dtmf transfering method;
9. Send and receive incoming caller id with q.931 protocol;
10. Supports calling directly to ip address;
11. Supports e.164 telephone number calling when gatekeeper used;
12. Search the gatekeeper in lan automatically;
13. Supports calling to pstn normal telephone with prepaid card issued by itsp ( net2phone, etalk etc.);
14. Supports pppoe (connect to internet with adsl and cable modem);
15. User can set the phone with standard web browser(internet explorer 6.0) or using telnet software;
16. Online software upgrade with ftp;
17. Supports g.723.15.3k/6.3k g.729, g.711a-law+law voice encoding and decoding arithmetic;
18. Dynamic voice inspection, cng (comfort noise generation) , dynamic voice jitter buffer;
19. According with g.165 16ms echo counteraction;
20. According with itu-t standard tone(dialing tone, busy tone etc.) and dtmf tone generation and inspection;
21. Supports the dialing rule basing on e.164 coding;
22. Up to 112 groups of fast dialing numbers;
23. 40 groups of incoming and outgoing numbers in memory;
24. 16 groups of voice message;
25. Volume of handset adjustable;
26. Volume of speaker adjustable;
27. Voice guide
1. Support SIP RFC3261 RFC3262 RFC3265, IAX2 protocal
2. Support G.711 / G723/G.729 codec
3. Support register two SIP and one IAX2 Servers simultaneously
4. Support PPPoE for XDSL
5. Web configuration through Built-in web server.
6. Support Bridge and both Router model
7. Support Voice Gain Setting, Jitter Buffer, VAD and CNG
8. Upgrading firmware and configurations by HTTP, FTP or TFTP
9. Power Over Ethernet or With headset funtion(Optional Parameters)
More...
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Voice signaling support sip simultaneous voip calls·+support sip v2 standard (rfc3261)·+register up to 4 server simultaneously support multiple dialing plan / call hunting group·+adaptive jitter buffer function·+support multiple dialing plan / call hunting group·+flexible routing table and profile·+extensible by external ivr/cdr/billing servers for value- added application
Management web interface management·+support auto-provision system·+remotely configuration/upgrade by web ui or auto provision build in watching dog for auto recovery
Call handing features call timer display call transfer, hold, forward, pickup·+local three-way conferencing·+one-touch speed dial for 6 keys missed call notification forward call (always/on busy/no answer) last call return redial·+call by ip address via dial pad voice message (extensible by external ivr) phone book directory (100 entries) ring melody selectable (voip/pstn/hold) customize melody·+time and date display
Dimensions:34cmx24cmx9cm(packing with box)
N.W.:1.8kgs g.W.:20kgs
Rfc 2833/inband dtmf
Stun - rfc 3489
Dhcp - rfc 2132
Http/tftp for provisioning and firmware upgrade
4. Advanced features
Wep 64/128 bit, wpa-psk tkip
Ap handover/roaming
Wifi network search
Auto scan
Sip server failover
Dns server failover
Voice mail indication
Aes encrypted remote provisioning
Scripting support for http based hotspot login
Web based provisioning
* requires network support.
Key features
Bar type design
802.11b/g
Sip based call processing
Web interface for management
Remote handset management
Earphone supported
Languages:english, spanish, french, chinese
Local log/system log
Technical profile
Subject specifications
Dimensions 11 x 4.5 x 2.2cm
(hxwxd)
Weight 111 g
Battery li-ion dc 3.6v
1500mah
Charger input 100~240vac
50 ~ 60hz 120ma
Talk time approximately 5 hours
(approx.)
Charging time 3 ~ 4 hours
Standby time 50 ~ 100 hours
Certification
Fcc id:06y-f1000g
Ic and ce
Systemstandards
Communication systemwi-fi (802.11b/g)
Frequency band 2.4ghz
Transmission output 20mw.
Device to be packed in box of each piece, 10 boxes to a carton.
Carton size:35*32*22.5cm
Gross weight:5.5kgs
Net weight:4.6kgs.
Product: voip phone
Model number: nxd-800
Voip phone function description:
Nxd-800 series voip phone is an internet based voice network phone terminal. Nxd-800 series voip phone 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.
Nxd-800 voip phone supports h323, sip, mgcp, iax2 and other protocols, offers one or two ethernet interface and is compatible with various softswitch systems and voip voice gateways to provide broadband ip voice service.
Voip phone (nxd-800)key features:
Unattended/ attended call transfer; call hold; configure file download and upload by palmtool.
Call waiting and per call-waiting blocking; adjustable user password and super password.
Call waiting and per call-waiting blocking; call forward (always, busy, no answer).
Voip speed dial; voip digital map; caller number/name display; support four settings storage, switch by keypad.
User define ring tone; e. 164 dial plan and customize dial rules; open source, user individuation firmware.
Voip speed dial; voip digital map; per area ring tone support; e. 164 dial plan.
Mwi(message wait indication); configurable/adaptive jitter buffer size.
Configurable hook flash time; static/dynamic wan-ip-addressing;
Supporting dhcp server, used for the dynamic address allocation plan of lan devices.
Static/dynamic wan-ip-addressing; pppoe; http, telnet, keypad and dedicate manage tool.
Dimensions:34cmx24cmx9cm(packing with box)
N.W.:1.8kgs g.W.:20kgs
Product :voip phone
Item:nxd-804 (support rohs and poe standard)
Description:
Dual voip/pstn interface ip-phone;easy to switch voip and pstn via ip-phone;
Voice signaling:support sip simultaneous voip calls;support sip v2 standard (rfc3261);register up to 4 server simultaneously; support multiple dialing plan / call hunting group;adaptive jitter buffer function;support multiple dialing plan / call hunting group;flexible routing table and profile;extensible by external ivr/cdr/billing servers for value- added application.
Management: web interface management;support auto-provision system;remotely configuration/upgrade by web ui or auto provision;build-in watching dog for auto recovery.
Call handing features:call timer display;call transfer, hold, forward, pickup;local three-way conferencing;one- touch speed dial for 6 keys;missed call notification;forward call (always/on busy/no answer);last call return redial;call by ip address via dial pad;voice message (extensible by external ivr);phone book directory (100 entries);ring melody selectable(voip/pstn/hold);customize melody;time and date display.
Dimensions:34cmx24cmx9cm(packing with box)
N.W.:1.8kgs g.W.:20kgs
Model number: NXD-801
Voip internet phone function description:
The Star-net voip internet phone NXD-801 series, which support MGCP, SIP, H. 323 Provides high quality voice communication over IP networks, It demonstrates the latest techn-ology and advancements in voip internet phone research, It is simplified operation and configuration are perfect for VoIP Service Provider, Enterprise and Home application. It has a two-port ethernet swith inside which makes it convenient for users to have both their IP Phone and PC connected to networks. NXD-801 supports Static IP config, DHCP Client and PPPOE client and it is suited for a lot different kinds of network enviroments.
Voip internet phone(NXD-801)Key Features:
1. Support H. 323 v4, compatible with most H. 323 v1-v4 system and devices;
2. Voip internet phone Built in H. 323 proxy support to pass NAT;
3. Voip internet phone Support MGCP RFC2705; Support SIP RFC3261; Support Net2phone private protocol;
4. Outband DTMF transmit by H. 245 user input or Q. 931 keypad; IEEE 802.3 /802.3 u 10 Base T / 100Base TX;
5. User define ring tone; E. 164 dial plan and customize dial rules; Open source, user individuation firmware;
6. DHCP support for automatically assign IP address and relevant parameters;
7. Support G.723.1 5.3k/6.3k, G.729, G. 711 A-Law, »-Law audio codec algorithm;
8. Tone generation and Local DTMF generation and detection according with ITU-T;
9. PPPoE support for ADSL or Cable modem; Voip internet phone Upgrade program by FTP mode.
10. Setting IP Net Phone parameters by standard web browser (such as IE6.0)phone keypad or standard telnet.
Product: voip phone
Model number: nxd-800
Voip phone function description:
Nxd-800 series voip phone is an internet based voice network phone terminal. Nxd-800 series voip phone 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.
Nxd-800 voip phone supports h323, sip, mgcp, iax2 and other protocols, offers one or two ethernet interface and is compatible with various softswitch systems and voip voice gateways to provide broadband ip voice service.
Voip phone (nxd-800)key features:
Unattended/ attended call transfer; call hold; configure file download and upload by palmtool.
Call waiting and per call-waiting blocking; adjustable user password and super password.
Call waiting and per call-waiting blocking; call forward (always, busy, no answer).
User define ring tone; e. 164 dial plan and customize dial rules; open source, user individuation firmware.
Voip speed dial; voip digital map; per area ring tone support; e. 164 dial plan.
Mwi(message wait indication); configurable/adaptive jitter buffer size.
Configurable hook flash time; static/dynamic wan-ip-addressing;
Supporting dhcp server, used for the dynamic address allocation plan of lan devices.
Static/dynamic wan-ip-addressing; pppoe; http, telnet, keypad and dedicate manage tool.
Dimensions:34cmx24cmx9cm(packing with box)
N.W.:1.8kgs g.W.:20kgs
JR-900 IP phone supports IAX2,SIP,protocols,Support Bridge and Router model.Support IAX2 and dual public server, offers Two Ethernet interface and is compatible with various softswitch systems and VoIP voice gateways to provide broadband IP voice service.
Features
Support SIP 2.0 (RFC3261) and correlative RFCs
Supprt IAX2
Codec:G.711A/u, G.7231 high/low, G.729, G.722
CP0020 is a DECT cordless IP phone that creates value for residential, traveler and business users. With DECT Cordless, PSTN and VoIP ability, users can not only make normal PSTN calls, but also take advantage of the features VoIP provides using SIP and H.323 protocol. Also noted is the ease of use with which the CP0020 operates, and the lightweight which allows for easy mobility.
JR-810 IP phone supports IAX2,two SIP,protocols,Support Bridge and Router model.Support IAX2 and dual public server, offers Two Ethernet interface and is compatible with various softswitch systems and VoIP voice gateways to provide broadband IP voice service.
Features
Support SIP 2.0 (RFC3261) and correlative RFCs
Supprt IAX2
Codec:G.711A/u, G.7231 high/low, G.729, G.722
Dimension(Wí-Hí-D)
200*160*80mm
Premium System Packages offer a stable IP PBX system embedded with 4 FXO ports and SIP open standards supporting up to 50 users. They are perfect for small to medium-sized business requiring rich features and high stability for an affordable price. Supporting both centralized and distributed telephony networks, these packages allow small-sized businesses to have a global communications presence and tremendous savings.
An IP phone system is a VoIP phone system that delivers voice traffic over the internet. IP phone systems are popular because they are less expensive than traditional phone systems and they offer a lot of features that traditional phone systems don't offer. IP phone systems are easy to set up and use and they are perfect for businesses of all sizes.
Key Features of the IP 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 (3.2í¦)
- 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.
Utstarcom gf210 gsm/wifi dualmode phone with gprs class 10 supported
Aside from the gsm and wifi dual-mode, the mobile phone only brings a set of basic features including a 1.8 inch 65k color display, text messaging, polyphonic ringtones, call forwarding, call waiting /hold /barring /conferencing, mms support, built-in games, a wap browser, alarm clock, organizer, notes, calculator, measure converter, world time and stop watch, so basically, everything a basic user can’t do without when considering to purchase a mobile phone.
Utstarcom gf210 additional features
• dual mode gsm/wifi dual mode phone
• gsm
O supports gsm triband networks
O 900/1800/1900 mhz
O limited support for 850 mhz
O voice codec: hr, fr, efr
O gprs class 10 supported
O wap 2.0*
O mms 1.2*
• wifi
O supports 2.4 ghz 802.11b
O auto scan/manual network selection
O effective range: up to 70m indoor, up to 120m outdoor
O protocols:
Sip, rtp/rtcp
Dhcp/static ip, stun, dns
Sip server failover
O voice process
Codec g.711a/’
Comfort noise generation
Voice activity detection
Adaptive jitter buffer
Echo cancellation
Inband/rfc2833 for dtmf
O security
Wpa-psk/wep (64/128 bit)
Supports 802.1x-md5/wpa2
O supports sip-message over wifi
O local/remote tftp/http firmware upgrade and provisioning
Utstarcom gf210 features
• display: 1.8’ 65k color cstn (128 x 160)
• call forwarding*
• call waiting/hold/barring/conference*
• ezi predictive text
• pim manager via syncml
• multilingual: english, chinese, german, spanish, french
• phonebook with 500 entries
• 40 polyphonic ringtones
• size: (h) 107 x (w) 45 x (d) 20 mm
• weight: 95 grams.
Packed in box of each piece, 10 boxes to a carton.
Carton size:35*32*22.5cm
G.W.:5.5kgs
N.W.:4.6kgs.
ò GSM Quad band: 850/900/1800/1900MHz ò WLAN 802.11b auto roaming support
ò Integrated VoIP over WLAN using SIP protocol on IMS system
ò Supporting active call handover between GSM and WLAN network
ò Form factor û Candy Bar : 112mm x 50mm x 12.5mm
ò Talk time û 3 hours (GSM); Standby û120 hours
ò 2.2ö QVGA 262K color TFT LCD 256K.
SIZE: 112mm x50mm x 12.5mm.
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