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ABTO VoIP SIP SDK Changelog

What's new in ABTO VoIP SIP SDK 4.7

Feb 18, 2015
  • Added new method GetSIPHeaderValueArr, returns array of values of specified header name.
  • Added ability to mute local video. See new method MuteLocalVideo
  • Improved method RetrieveExternalAddress. Now it returns IP and port.
  • Added new property AECDelayMs
  • Fixed possible crash when received incoming call during SDK initalization

New in ABTO VoIP SIP SDK 4.5 (Aug 22, 2012)

  • Fixed playing dial tones
  • Added new method GetSIPHeaderValueLine
  • Added ability to use method GetSIPHeaderValue on outgoing messages too
  • Added ability to handle NOTIFY messages event when SDK is not subscribed for them.
  • Added ability to record microphone sound only. StartRecordingConnection(0);
  • Fixed wrong G729 codec file in installation package
  • Added ability to use fixed transport in Via header
  • Added ability to automatically detect sound output device channel count
  • Fixed bug with ‘From’ header in outgoing text messages
  • Changed SDK version in dll resources
  • Added ability to change ringtone depending on the incoming call
  • Added advanced log on selection camera format and selection source interface
  • Fixed selection source interface in WinCompat::determineSourceInterface
  • Added new feature ‘muLtiple sip accounts registration’
  • Added ability to enable\disable video call feature without restarting phone

New in ABTO VoIP SIP SDK 4.0 (Nov 10, 2011)

  • Added ability to set CallInviteTimer. Default value is equal to 40 seconds.
  • Disabled playing local dial tones to remote side.
  • Added ability to handle “Registration removed” event.
  • Additional fix of ’183+SDP’ message handling.
  • Added SRTP support.
  • Updated help documentation.
  • Redesigned code that checks RTP port range availability.
  • Added new codecs codec_amrwb.dll and codec_amrnb.dll.
  • Added ability to send/receive text messages to all examples.
  • Redesigned example with right sound options name (echo, noise, autogain).
  • Fixed bug with Hold/Retrieve conference call.
  • Modified code that handles receiving UPDATE sip message.
  • Added ‘UseMixerPlayer’ setting.
  • Removed previous changes, the SDK plays files depending on this option.
  • Modified code that handles receiving messages 180 Ringing with enabled 100rel.
  • Added ability to send PRACK answer.

New in ABTO VoIP SIP SDK 3.0.12 (Nov 10, 2011)

  • Added ability store ini file in utf-16 format, load/store ini files from “Settings” form
  • Improved applying new settings procedure. It allows avoid bugs with changing registration/network settings in runtime
  • Added new example MlSampleSkinCPP, Allows create and apply new skins without changing source code, like Winamp
  • Added ‘AutoAnswer’ feature
  • Àutomatic discovering available network interfaces and selecting better to reach registrar server.
  • Manual selection of network interface for RTP data exchange.
  • Redesigned STUN feature
  • Added feature to detect availability SIP/RTP ports
  • Added ability to send/receive text messages
  • Added fax tone detection
  • Added ability to display CallerId in generated events
  • Added ability to display call duration
  • Added ability to change network interface to VB6, VB.NET examples

New in ABTO VoIP SIP SDK 3.0.11 (Nov 10, 2011)

  • Added ability store ini file in utf-16 format
  • Renamed to “ABTO SIP SDK”
  • Added ability load/store ini files from “Setting” dialog in MlSampleWindowCpp
  • Redesigned changing settings procedure. It allows avoid bugs with changing registartration/netowork settings in runtime
  • Added new example MlSampleSkinCPP
  • Added ability to set network interface to CS and Delphi examples
  • ‘Run now’ options starts new exmaple MlSampleSkinCPP
  • Added ‘AutoAnswer’ feature
  • Fixed Ml-VB6 example bug with renamed dll
  • Fixed bug in condition that checks is open SIP port.
  • Added ability to change network interface to VB6, VB.NET examples

New in ABTO VoIP SIP SDK 3.0.6 (Nov 10, 2011)

  • Component use first found IP V4 by default
  • Manual network interface selection
  • New examples:
  • 1. “MlSampleGSMGateway” that allows route incoming calls to GSM phone though bluetooth
  • 2. “MlSampleOperator” Can be used as call center operator terminal. Made as auto-answer phone Allows for manager connect to operator and listen conversation

New in ABTO VoIP SIP SDK 3.0.5 (Nov 10, 2011)

  • AuthID handling
  • New config variable “LocalIP”

New in ABTO VoIP SIP SDK 3.0.4 (Nov 10, 2011)

  • Explicit multilines handling added
  • Conferences quality issues fixed
  • Configuation handling enhanced
  • Stability has been greatly improved
  • New samples added
  • Help updated
  • Samples updated

New in ABTO VoIP SIP SDK 2.1.4 (Nov 10, 2011)

  • Added FrameReceived event
  • Saving codec checked state when changing codec order
  • Added .Net wrapper
  • Added C++ Windowless Sample.
  • Help updated
  • Samples updated

New in ABTO VoIP SIP SDK 2.1.3 (Nov 10, 2011)

  • Added DLL only version, along with ActiveX
  • C# sample for DLL version
  • C# Console version updated
  • C++ application update
  • SDP updates
  • Conference calls updated
  • Caller-id and user-agent customization
  • Retrieving external IP addres from STUN server

New in ABTO VoIP SIP SDK 2.1 (Nov 10, 2011)

  • IM interface added
  • fixed issue with RFC4566
  • Codec disabling
  • Help updated
  • Samples updated
  • Playback of different formats of WAV files
  • Software volume control

New in ABTO VoIP SIP SDK 2.0 (Nov 10, 2011)

  • Noise reduction
  • Auto gain
  • Added local number parameter
  • Help updated
  • Jitter buffer parameters
  • Samples updated
  • Windowless samples on C++ and .NET

New in ABTO VoIP SIP SDK 1.3 (Nov 1, 2010)

  • 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
  • Additional:
  • Fixed: Blind (unattended) transfer and Consultative (attended) transfer
  • New methods such as get_CallState, get_LineResponseCode, get_LineResponseText, get_LocalIP, get_NegotiatedCodecName, get_NegotiatedPayloadType, get_NetworkAdapter, get_RemoteURI, URLGetAOR
  • New properties such as DnsSrvEnabled, MinPrefetchCount, MaxPrefetchCount, MaxRTPPackets, RegistrationResponseCode, RegistrationResponseText
  • New events: OnIncomingSipRequest, OnIncomingSipResponse, OnOutgoingSipRequest, OnOutgoingSipResponse)
  • New events: OnLocalMediaStarted, OnLocalMediaStoped, OnRemoteMediaStarted, OnRemoteMediaStoped)
  • New, re-written and updated samples
  • MS IE Browser (webdemo), VB6 bug fixed
  • Fixed: License issue on Windows 2000 and web licenses

New in ABTO VoIP SIP SDK 1.1 (Oct 6, 2007)

  • The VoIP SIP 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.