Grandi IP phone, ATAs and VoIP gateway

Wed, 08 Nov 2006 15:36:24 +0100

Vendor:
Grandi Digital Information Ltd.

Product URL:
http://www.easy3call.com/products/index.htm

Product image:

HW/OS Platform:
ARM 9

Comment:

The Grandi GIP300 IP phone is a new entry level IP phone. GIP300 has dual 10/100M Ethernet ports that can be configured as either switched or routed ports, built-in NATi router/DHCP server, high quality full duplex hands-free speakerphone with acoustic echo cancellation, headset jack, message waiting indicator, and more memory for future function growth. It supports G.711i, G.723.1i, G.729A/B. GIP300 is able customer to register two SIP servers simultaneously. In addition, you could download music ringing tones and the GIP300 provides as many as eight kinds of ring tongs.

The Grandi GIX100 VoIPi ATAs works great. It is very easy to configure, excellent audio and great selection of codecs including G.711, G.723.1, G.729A/B. It is inexpensive with dual 10/100M Ethernet ports that can be configured as either switched or routed ports, built-in NAT router/DHCP server and support message waiting indicator. And most importantly, GIX100 is an ultra-affordable VoIP ATAs with high performance.

Both GIP300 and GIX100 is a new generation VOIP terminal products which were developed based on single ARM9E CPU platform. Its core is a ARM9E CPU at 150MHZ. With this technology, Grandi would provide the products with high performance and ultra-affordability cost.

GIP300 is just at $30.00 USD for 100K units order.
GIX100 is just at $28.00 USD for 100K units order.

E-mail:
voipsales@grandi.cn

iistr phone

Mon, 19 Dec 2011 01:32:18 +0100

Vendor:
levent

Vendor's URL:
iistr.net

Product URL:
iistr.net

test

Tue, 08 Nov 2011 14:43:34 +0100

Vendor:
test

Vendor's URL:
test.com

Product URL:
test

E-mail:
xnbnl0@gmail.com

voxalot fher

Wed, 26 Oct 2011 02:01:10 +0200

Vendor:
us.voxalot.com

Vendor's URL:
www.voxalot.com

Product URL:
www.voxalot.com

Comment:
voip

E-mail:
fernando_sanchez@ejecutivos.com

karl

Tue, 23 Aug 2011 04:44:04 +0200

Vendor:
iptel.org

Vendor's URL:
http://www.iptel.org

Product URL:
http://www.iptel.org

satoshun

Mon, 18 Jul 2011 09:20:49 +0200

Vendor:
satoshun

Vendor's URL:
google

Product URL:
google

WeSoFly Entertainment Phone

Mon, 18 Jul 2011 05:34:15 +0200

Vendor:
WeSoFlyEntertainment.com

Vendor's URL:
www.WeSoFlyEntertainment.com

Product URL:
http://www.wesoflyentertainment.com/member/

E-mail:
corporate@WeSoFlyEntertainment.com

iptel

Thu, 07 Apr 2011 09:24:46 +0200

Vendor:
iptel

Vendor's URL:
iptel.org

Product URL:
iptel.org

Frank

Fri, 25 Mar 2011 07:53:07 +0100

Vendor:
Frankieboy

Vendor's URL:
www.mastering.fi

Product URL:
www.mastering.fi

E-mail:
timo.raunio@netikka.fi

sip

Fri, 11 Mar 2011 01:48:50 +0100

Vendor:
hurri

Vendor's URL:
iptel.org

Product URL:
iptel.org

lhl7998

Thu, 02 Dec 2010 14:59:19 +0100

Vendor:
lhl7998

Vendor's URL:
sip.iptel.org

Product URL:
sip.iptel.org

IMSDroid

Fri, 26 Nov 2010 02:20:16 +0100

Vendor:
Idk

Vendor's URL:
Idk.com

Product URL:
idk.com

VOIP SIP SDK

Tue, 07 Sep 2010 10:39:03 +0200

Vendor:
VOIP SIP SDK

Vendor's URL:
http://ipphonesdk.com

Product URL:
http://ipphonesdk.com

HW/OS Platform:
Win 2000/XP/Vista

Comment:

SIP Client SDK provides a powerful and highly versatile solution to add quickly SIP (Session Initiation Protocol) based dial and receive phone calls features in your software applications and websites. It accelerates the development of SIP/RTPi compliant soft phone with a fully-customizable user interface and brand name.

The VoIPi SIP Client SDK contains a high performance VoIP conferencing client capable of delivering crystal clear sound even for both low and high-bandwidth users and SIP compatible devices (hardware and software). It enables a worldwide communication over the internet or intern networks either and delivers superior voice quality by integrating digital voice processing features including auto gain controller (AGCi), acoustic echo suppression (AES) and noise suppression. It supports multiple lines, multi-party voice conference, call hold, call forwarding and transfer, DTMFi, Packet Loss Concealment (PLC), adaptive jitter buffer, record and playing WAV and much more!

VoIP SIP Client SDK is based on IETFi standards (SIP, RTP, STUN, TURN, ICE etc.), so it should be compatible with other standard based products such as: SERi, Sip EXpress, OpenSER and Asterisk.

New features of the VoIP SIP Client SDK:

• g729 and g723 Codeci´s support
• Multiple and single Codec selection support
• Failure codes support (get SIP Message Response Code, SIP Message Response Text)
• RTP/RTCPi Port setting (for inbound RTP traffic)
• Reduce audio latency and audio latency settings (properties: MinPrefetchCount, MaxPrefetchCount, MaxRTPPackets)
• Media status (Events: OnLocalMediaStarted, OnLocalMediaStoped, OnRemoteMediaStarted, OnRemoteMediaStoped)
• Get used codec per line
• Custom Ringtone (play wav) support (property: RingtoneFile)
• Play wav to a selected phone line (methods: StartPlayingAtLine, StopPlayingAtLine)
• Redirect Call to other phone line
• Load and Save Configurations (methods: LoadConfiguration, StoreConfiguration)
• Complete new, re-written and updated samples with source code
• and much more!

Here is a list of the main features of the VoIP SIP Client SDK:

• Easily make and receive SIP (Session Initiation Protocol) based phone calls through any    SIP gateway or SIP compliant IP-Telephony service provider
• VoIP conferencing with crystal clear sound even for both low and high-bandwidth users
   G711 A-Law, G711 U-Law, Speex, Speex-wb, GSM6.10, iLBCi, L16 and g729 & g723 Codec
• Open standards-based and interoperable with all of the major equipment vendors
• UDP and TCP support for sip activex
• Multi-party voice conference support/ Conference split and join, locally mixed conferences
• Multi-line support (multiple simultaneous calls)
• SIP Instant/Chat Messaging with send/receive controlling with direct sip activex support
• Integrated STUN, TURN and ICE support
<• Comes with new sample SIP Proxy Server    to provide in bundle with the SIP    Client ActiveX a ready up own SIP VoIP    and Instant Messaging network solution.
• P2P support for directly connections    between 2 SIP clients( sip activex ) without SIP Server
• Outbound proxy server support
• Encrypted SIP account settings (encrypted    SIP account settings in your webpage)
• Line Hold/Un-hold support
• Call forwarding and rejection
• Call transfer support
• Select media input/output devices
   on-the-fly - also during a conversation/    conference)
• Mute microphone/speaker + level indicator
• Auto-answer
• DNDi (Do Not Disturb)
• Adaptive Jitter buffer
• PLC (Packet Lost Concealment)
• AGC (auto gain controller)
• AES (Acoustic echo cancellation or    suppression)
• Noise cancellation or suppression
• DTMF tones support (generation/detection)
• Recording voice conversation into PCMi    WAVE (.wav) file
• Playing PCM WAVE (.wav) files to the    remote end
• Audio file memory cache
• Extended SIP URL functions
• Dynamically loadable codec support (coming soon)
• Comes as ActiveX control (Web demo with ready-up signed CAB included)
• Registration on SIP Server (SIP Registrar)
• Log file on/off setting
• Microphone and Speaker Volume with Mute support
• Keep-alive packets to NATi/firewall
• Fully-customizable user interface
• Microsoft Authenticode Certificate
• Works with all kind of Internet connections
• Friendly to NAT and other firewalls
• Keep-alive packets to NAT/firewall
• Royalty free licensing
• No Yearly/Monthly fee
• Very easy to incorporate
• Fully sample applications for various programming languages such as sample source code    for C#, VB.NET, JavaScript (Webdemo), VB 6.0 and Delphi
• For .NET framework as well and all development environments with ActiveX support

Easy, familiar, event-driven call control ActiveX
• Easy to use; quick development
• Support for .NET framework and all development environments with ActiveX support
• Very easy to incorporate

Rich call control feature set
• Multi-party voice conference support (Conference split/ join, locally mixed conferences)
• Multi-line support (multiple simultaneous calls)
• SIP Instant messaging
• Locally mixed conferences
• Hold/Mute
• Call transfer
• Call forwarding and rejection

Industry leading SIP support
• RFC3261 compliant SIP stack
• RFC 2833 out-of-band DTMF signaling
• Integrated STUN, TURN and ICE support

Comprehensive configuration support
• Select media input/output devices (on-the-fly as well during a conversation/conference)
• Configurable ports (RTP, SIP UDP, SIP TCP, STUN, TURN, ICE)
• SIP proxy

Advanced digital voice processing features
• AGC (auto gain controller)
• AES (Acoustic echo cancellation or suppression)
• Noise cancellation or suppression

… and much more!

E-mail:
support@sipvoipsdk.com

Llamadas Gratis

Wed, 21 Jul 2010 03:14:33 +0200

Vendor:
Doddle Phone

Vendor's URL:
www.paypallatino.com

Product URL:
www.paypallatino.com/visa.php

.NET SIP Library

Thu, 15 Jul 2010 07:58:20 +0200

Vendor:
Konnetic

Vendor's URL:
www.konnetic.com

Product URL:
http://www.konnetic.com/products/products_sip_sdk_std.aspx

HW/OS Platform:
.NET/Mono

Comment:
A set of RFC-compliant high-functionality class libraries which manage SIP messaging. Designed to support the development of cross-platform multimedia and text-based applications and services written any any .NET language (e.g. c#, VB or F#).

E-mail:
james@konnetic.com

iptel.org - SIP