Question:
Software Product: conaito VoIP SIP Client SDK - Version: 1.3
Author: conaito Technologies
Web: http://www.conaito.com
Email: support@conaito.com conaito VoIP SIP Client SDK
A powerful and highly versatile VoIP SDK to accelerate development of SIP applications and websites Our brand-new 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/RTP compliant soft phone with a fully-customizable user interface and brand name. The conaito VoIP 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 (AGC), acoustic echo suppression (AES) and noise suppression. It supports multiple lines, multi-party voice conference, call hold, call forwarding and transfer, DTMF, Packet Loss Concealment (PLC), adaptive jitter buffer, record and playing WAV and much more! conaito VoIP SIP Client SDK is based on IETF standards (SIP, RTP, STUN, TURN, ICE etc.), so it should be compatible with other standard based products such as: SER, Sip EXpress, OpenSER and Asterisk. New features of the conaito VoIP SIP Client SDK - version 1.3: g729 and g723 Codec΄s support
Multiple and single Codec selection support
Failure codes support (get SIP Message Response Code, SIP Message Response Text)
RTP/RTCP 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 conaito 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, iLBC, L16 and g729 & g723 Codec)
Open standards-based and interoperable with all of the major equipment vendors
UDP and TCP support
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
Integrated STUN, TURN and ICE NAT Traversal
Comes with new sample SIP Proxy Server to provide in bundle with the conaito SIP Client ActiveX a ready up own SIP VoIP and Instant Messaging network solution.
P2P support for directly connections between 2 SIP clients 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 (as well during a conversation/conference)
Mute microphone/speaker (including level indicator)
Auto-answer
DND (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 PCM 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 NAT/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 as well and all development environments with ActiveX support
Very easy to incorporate Rich call control feature set Multi-party voice conference support (Conference split and join, locally mixed conferences)
Multi-line support (multiple simultaneous calls)
SIP Instant messaging
Locally mixed conferences
Hold/Mute
Call transfer and redirect
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! Having the above features available makes it simple to develop any type of VoIP-enabled application, like e.g. a SIP softphone, IVR solution, teaching tool, live support, voip chat, meeting tool or any other type of application which requires users being able to talk and type messages to each other. For conaito VoIP SIP clients to be able to interact with each other they must connect to a SIP gateway or SIP based IP-Telephony service provider. Just relax!
Please, don't hesitate trying our VoIP SIP Client SDK at once and get yourself, as well as your customers, the exciting experience of easy, fast and high quality standard applications which VoIP-enable your application and website. We hope you enjoy the new conaito VoIP SIP Client SDK A powerful and highly versatile VoIP SDK to accelerate development of SIP applications and websites.
conaito Technologies
http://www.conaito.com
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