conaito VoIP SIP SDK 3.0 launches

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conaito VoIP SIP SDK 3.0 launches

conaito sipsdk.jpgconaito has an interesting VoIP SIP SDK for developing SIP applications for websites. Starting today, the new VoIP SIP SDK v3.0 is available to download. conaito VoIP SIP SDK is based on IETF standards (SIP, RTP/RTCP, STUN, TURN, ICE, etc.), so it should be compatible with SER, Sip EXpress, OpenSER, Asterisk, and other popular SIP-based solutions.

New features of the conaito VoIP SIP client - version 3.0 include:

  • ICE Support
  • TURN Server support (exp.)
  • VAD (Voice Activity Detection)
  • IPv6 support (exp.)
  • Keep-Alive messages PLUS Interval setting
  • Encrypted SIP Settings (protect your SIP account settings in websites)
  • In-band/out-band DTMF
  • Play DTMF tones (local maschine)
  • Playing Ringtones (local maschine)
  • Call Transfer (PLUS Transfer with Replaces)
  • Conversation/Conference recording to MP3 file
  • Modify outgoing SIP messages
  • Sound device detection
  • Enable/Disable sounds devices
  • Set Online Status (RFC 4880)
  • SIP Traces for RX and TX
  • Set Media Quality - the lower value is the lower quality is reached (but better performance)
  • Set RTP packet droping (0% - 100%) - Rx and Tx
  • New ready-up "how-to-start" - Samples
  • Complete new re-written samples with source code

Here is a list of the main features of the conaito VoIP SIP client:
• 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, GSM6.10, iLBC, L16, g723 and g729 Codec)
• Encrypt SIP account settings (protect your SIP account settings in websites)
• Secure Weblicensing (protect your license in websites)
• Multi-User conference support
• Multi-line (simultaneous calls) support (Multiple Concurrent calls)
• Call Hold support
• Call Transfer support (PLUS Call Transfer with Replaces)
• Instant text messaging (MIME) support and typing indication
• Mute microphone/speaker for each line
• DNS SRV resolution for SIP servers (RFC 3263)
• Stereo codec (L16)
• Auto-answer
• Do Not Disturb (DND)
• Adaptive jitter buffer
• Adaptive silence detection
• PLC (Packet Lost Concealment)
• DTMF tones support (in-band/out-band)
• Recording voice conversation into PCM WAVE (.wav) and MP3 file
• Playing PCM WAVE (.wav) files to a voice conversation
• Dynamically loadable codec’s support (g723, g729 codec's plug-in samples included)
• Comes as ActiveX control (Web demo with ready-up signed CAB included)
• Registration on SIP Server (SIP Registrar)
• Support UDP and TCP as transport type
• Instant messaging (Message)
• Microphone and Speaker Visualization support
• Microphone and Speaker Volume with Mute support
• Audio device selection and detection
• Fully-customizable user interface
• Microsoft Authenticode Certificate
• Works with all kind of Internet connections
• Royalty free licensing
• No Yearly/Monthly fee
• Very easy to incorporate
• Fully commented sample applications for various programming languages
• Sample source code for C#, VB.NET, JavaScript (Webdemo), C++, VB 6.0 and Delphi 7.0
• For .NET framework as well and all development environments with ActiveX support

Via: conaito blog.

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