SIP

Grandi IP phone, ATAs and VoIP gateway

easy3call on 8 November, 2006 - 14:36
easy3call's picture
Keywords: Gateway | H.323 | Hardphone | SIP

Grandi Digital Information Ltd.

www.easy3call.com

Grandi IP phone, ATAs and VoIP gateway

ARM 9

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.

teste

inunessip on 25 March, 2014 - 16:28
Keywords: SIP

teste

teste

teste

test

teteren on 24 March, 2014 - 11:36
Keywords: SIP

test

http://www.iptel.org/

http://www.iptel.org/

Ibrar

ibrar630 on 1 March, 2014 - 13:47
Keywords: SIP

ibrar

ibrar

ibrar

test bigdeit

hoang on 20 February, 2014 - 08:02
Keywords: SIP

dfghj

xnegru on 28 January, 2014 - 12:48
Keywords: SIP

vbnjhjk

www.dtrf.com

www.dtrf.com

test

acharya on 10 November, 2013 - 19:55
Keywords: SIP

Jumblo

www.jumblo.com

www.jumblo.com

cat9999

cat9999sss on 3 November, 2013 - 11:16
Keywords: SIP

cat9999

9527.ro

9527.ro

windows

asdasdasd


85153331@qq.com

MY PHONE

tontojohn on 14 October, 2013 - 13:16
Keywords: SIP

ipkall

www.ipkall.com

www.nutradietlean.com

WhichVoIP SIP Trunking

hollibenjamin on 27 August, 2013 - 15:30
Keywords: SIP

WhichVoIP

http://www.whichvoip.com

WhichVoIP is an informational site with voip and sip provider comparisons and helpful articles

SIP BNR

sipbnr on 14 August, 2013 - 10:28
Keywords: SIP

Jitsi

https://jitsi.org/

https://jitsi.org/

SIP/E1 Gateway - TDM Switch supporting SIP protocol

terratel on 15 February, 2013 - 08:14
Keywords: Gateway | SIP

TERRATEL

TERRATEL provided SIP/E1 Gateway, a digital TDM switch supporting SIP (compatibility with the Asterisk, FreeSWITCH, SoftSwitch etc.), is a universal product designed to provide VoIPi and TDM telephony solutions. The switch is aimed at optimizing solutions for building and upgrading corporate networks by applying open Asterisk/FreeSWITCH/SoftSwitch -based IP telephony solutions. The figure shows the block diagram demonstrating the possible application of the switch.

Basso Puhelin

Basso on 17 January, 2013 - 11:15
Keywords: SIP

Basso Puhelin

basso.fi

basso.fi

call

loverswinter on 9 July, 2012 - 03:54
Keywords: SIP

IPKall

MySipPhone

aftad on 8 March, 2012 - 08:47
Keywords: SIP | Softphone

Lumais Soft.

Windows

MySipPhone SIP/VoIPi application for Windows OS.

Features:
1. SIP Voice and Video call
2. STUN support for NATi traversal
3. Multiple codec selection and ordering of codecs
4. Blind call transfer
5. Call transfer with consultation (attended call transfer)
6. Call reject
7. Call Hold
8. System tray icon


Audio and Video Codecs:
1. audio: G.711i-ALaw-64k

levent

iistr.net

iistr.net

voxalot fher

axxessotel on 26 October, 2011 - 00:01
Keywords: SIP

us.voxalot.com

www.voxalot.com

www.voxalot.com

voip

VOIP SIP SDK

itspecialist on 7 September, 2010 - 08:39
Keywords: SIP

VOIP SIP SDK

http://ipphonesdk.com

http://ipphonesdk.com

Win 2000/XP/Vista

 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!

.NET SIP Library

JamesWright on 15 July, 2010 - 05:58
Keywords: Protocol Stack | SIP

Konnetic

www.konnetic.com

.NET/Mono

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#).

Linphone

TJipson75 on 15 June, 2010 - 17:02
Keywords: SIP

Simon Morlat

http://simon.morlat.org

www.linphone.com

Android

Low cost SIP/IAX2 IP phone support router, VPN, auto provisioning

sophiazxy on 29 October, 2009 - 03:19
Keywords: SIP

Wuchuan Network (Shenzhen) Limited

www.5111soft.com

Support Protocol

◆ SIP (RFC3261, RFC2543)
◆ Support IAX2 protocol
◆ Support codec: G.711A/u, G.722,G.723 high/low, G.729i A/B
◆ Support G.168 echo cancellation standard, compliant 96ms echo cancellation with speaker mode
◆ Support voice volume adjustment, including IN/OUT of handset and speaker
◆ Support Jitter Buffer, VAD, CNG, SIP, Domain name register, point-to-point Call
◆ Support RTPi and RTCPi
◆ Support the Inbound/Outbound transmission; SIP info, DTMFi Relay, RFC2833
◆ Support many countries' standard ring
◆ Support NATi: Support STUN, CITRON, AVS Mode
◆ Support SIP domain, SIP Authentication (none, basic, MD5), Domain Name parse
◆ Support two SIP server synchronously, including Pubic Server/ Private server, can make a call by any proxy. You can back-up and select any above SIP server.
◆Support SIP application, including SIP Call forward/transfer/holding/conference/pickup/redial/unredial/joincall


Wuchuan Network Tech Co.,Ltd

http://www.5111soft.com

Hello fellows,

I'm Sophia from Wuchuan Network (Shenzhen) Ltd. Info below is about our best sale product for now---SIP/IAX2 VoIPi Phone PH802.which is a unique SIP Phone featuring for IP PBXi.Its brand is Wuchuan or Neutral. If you are interested in our product, pls don't hesitate to contact me.

Wuchuan Network Tech Co.,Ltd

http://www.5111soft.com

Hi friends,

I'm Sophia from Wuchuan Network (Shenzhen) Ltd. The following Info is about our newly high tech product --SIP/IAX2 VoIPi Phone PH806 -- A newly developed hi-tech SIP phone for enterprise supporting 5 SIP servers synchronously,(POE) function and 128*64 LCD display.Besides, it supports 3 interactive soft key to play humanized operating prompt and an option of a pair of headsets for private calls.

Our PH806 has all basic functions that other IP phones have, and belows are our phone’s special and advanced functions:

Wuchuan Network (Shenzhen) Limited

http://www.5111soft.com

IP Phone PH217 completely follows VOIP standard offered by ISO, setting in two Protocols: SIP, fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software in the market.Simple IP phone for home users

Start to enjoy a IP Phone with online status display function as ICQ or MSN!

viva0101 on 1 September, 2009 - 07:21
Keywords: SIP

Wuchuan Network (Shenzhen) Limited

http://www.5111soft.com

The two IP Phones are our best sale products for now.
The unique phones can display directly the online state of the booking numbers by the indicator. If the indicator is green, means it is online; the indicator is green and twinkle, means it is in the course of the call; if red, means it is offline.

Wuchuan Network (Shenzhen) Limited

http://www.5111soft.com

 A newly developed hi-tech SIP phone for enterprise supporting 5 SIP servers synchronously,(POE) function and 128*64 LCD display.Besides, it supports 3 interactive soft key to play humanized operating prompt and an option of a pair of headsets for private calls.

Our PH806 has all basic functions that other IP phones have, and belows are our phone’s special and advanced functions:

--Support IAX2
--Support 5 SIP servers and 1 IAX2 account synchronously, can call in and out by either proxy
--Support two models: Bridge and Router, integrate two ports router function.

Wuchuan Network (Shenzhen) Limited

http://www.5111soft.com

Our PH802 is an award-winning next generation IP phone supporting SIP/IAX2 and online status display.Featuring for VPN,VLAN,superb sound quality and rich functionalities at ultra-affordable price,it is a special phone for IP PBXi,can display presence status.

Our PH802 has all basic functions that other IP phones have, and belows are our phone’s special and advanced functions:

--Support IAX2.
--Support two SIP server synchronously, including Pubic Server/ Private server,can make a call by any --proxy. You can back-up and select any above SIP server.

--Support BLF, server presence and Peer to peer presence negotiation
--Support VPN (L2TP) clients
--Support two models: Bridge and Router, integrate two ports router function.
--Support basic NATi and NAPTi
--Support PPPoE for xDSL, and support auto redial when disconnect
--Support SNTP Client to get time from internet
--Support second layer QoS (802.1p)
--Support VLAN
--Support VPN, L2TP protocol. (New hardware support openVPN)
--With IP PBX,support 10 group quick dial number, together with IP PBX presence subscribe, the IP phone can display directly the online state of the booking numbers by the indicator. If the indicator is green, means it is online; the indicator is green and twinkle, means it is in the course of the call; if red, means it is offline ——only our phone can make it !
--Support the local voice message, play the message by one key, IVRi personality record the message, voice prompt.
--Caller ID display, ban calling out, avoid-disturb setting, auto-answering, auto dial while picking up the telephone, quick dial;
--Support Silence Suppression, VAD (Voice Activity Detection)
--Support local/server Message-Waiting Indication
--Support auto long-distance configuration and edition auto-upgrade.
--Support call record checking and management.
--Support join Call
--Support directly dial IP+port to call SIP terminal device

ABTO SIP VOIP SDK

Alex Muzychuk on 15 May, 2009 - 13:20
Keywords: H.323 | PBX | SIP | Softphone

ABTO LLC

http://www.abtollc.com

Win 2000/XP/Vista

Mercuro IMS Client

bossiel on 17 September, 2008 - 13:29
Keywords: SIP | Softphone

Inexbee

Mercuro IMS Client

Windows 2000/XP/Vista

The Mercuro IMS Client is fully compliant with the main 3GPP IMS specifications and partially compliant with RCS (Rich Communication Suite) phase 1 and OMA specifications.

GP-710 Bluetooth VoIP Gateway

gempro on 12 March, 2008 - 06:54
Keywords: Gateway | SIP

Gempro Technology Inc.

http://www.gempro.com.tw

GP-710 Bluetooth VoIP Gateway

[GP-710] is a revolution and innovation product.
It Integrated VoIPi and 2.4G Bluetooth technical in a device,
this not only offered high cost-effectiveness in mobile communication for mobile user,
but also offered very flexibility in re-building system.

Feature:
1. Could use with GSMi/CDMA/WCDMA/3G/3.5G
    various Bluetooth mobile phone.
2. VoIP and Bluetooth Mobile fully integration.
3. Compatible with SIP RFC543, RFC3261.
4. Support UPLINK , DOWNLINK Routining.
5. Support Voice report and Setting IP function
6. Offered Web Site for inquire or setting.
7. Support SIP PROXY, or point to point     application.
8. Offered one stage ,two stage free dialing     and call transfer function.
9. Bluetooth with auto pairing and auto     searching function.
10.QoS and Digital Transmit.

Specification:
VoIP
Web Browser
IVRi Interface
Uplink Route Setting
Downlink Setting
1 Stage, Dialout Called, Free Dial

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