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You can download the latest version of Ozeki SDK by clicking on the link below. After download, you need to unzip it and run the installer in the zip package.
Ozeki SDK v10.4.54Updated: 2024.09.18.
- Improvement (Google TTS):From this version the Google TTS request is sent over a HTTPS protocol by default. A UseHttps variable was introduced, which can be used to switch back to HTTP.
- New feature (Font Size): When a text overlay is used above a video stream, the font size of the text and the position of the text can now be adjusted using attributes in a very easy way
- Improvement (RTSP stream): If multiple network interfaces, such as VPN links, ethernet cards or wifi network adapters are present, the IP address of the IP camera stream is determined by the ip address assigned to the interface used for outbound traffic. This makes the RTSP stream work reliably on complex network setups
- Bug fix (Common Dependency Resolver): When the SDK dll was installed in the Windows Global Assembly Cache, the applications based on it have loaded the dll twice. This resulted in a CommonDependencyResolver exception. This issue has been solved.
- Improvement (Better device management): When you request access to a media device, such as a Speaker or Microphone, the same (single) instance of the device class is returned, because the same hardware device is accessed. If two users (e.g. John and Bob) use the same hardware, their activity interact: for example John may start and Bob may stop the device. If John starts the Microphone and Bob stops it, the microphone will stop at John as well.
In this new version we have introduced new methods: StartManaged/StopManaged/DisposeManaged. If you use these new methods, the two users will not effect each other. So if you have two or more users accessing the same media devices swich to StartManaged/StopManaged/DisposeManaged to get better functionality.
- Bug fix (INVITE TIMEOUT): On some systems the SIP INVITE message response parsing was invalid. This resulted in an INVITE TIMEOUT error messages. This error was related to the introduction of new hashing algorithms to support better security. The issue is resolved in this release.
- Update (.NET Framework): We have updated the minimum .NET Framework requirement of Ozeki SDK to .NET Framework 4.8. The previous version was .NET Framework 4.0. The update was necessary to support several projects and to resolve some conflicts that arose under Visual Studio 2022
- New feature (sha 256): The SHA256 authentication algorithm is now supported.
- New feature (StartManaged/StopManaged): If multiple threads want to access the physical audio/video devices in the system (for example the speaker), they should use the StartManaged/StopManaged and DisposeManaged procedures instead of Start/Stop/Dispose. These procedure make access to these devices thread safe, and ensures that multiple threads can use them without issues. These managed functions make sure, that the speaker will be started/stoped/disposed only after all threads have completed their operations. Note, that these functions are based on weak references, so it is not a problem if a developer forgets to call the dispose method.
- Improvement (Audio and video device detection): To detect new audio/video devices plugged into the system you can use the built in functions of Windows. In this version, the Ozeki library will allow you to query the plugged in device list unlimited times in a thread safe mode. This enables you to work with dinamically attached and removed devices more efficiently.
- Bug fix (Blind transfer): The blind transfer had a bug. It did not parse the target field properly in all cases. This bug is fixed.
- Improvement (SIP Messaging): The SIP/VoIP Instant messaging features has been improved. The previous versions contained a bug. The addresses of the instant messages were not set properly. This problem is fixed in this version.
- New feature (OnForwarded): This new event was added to the softphone to make it possible to handle SIP 181 redirect events. This event is sent by the PBX to the softphone in case the call is redirected. Asterisk and many other SIP PBX systems support this event.
- Improvement (SRTP): The SRTP audio and video stream did not work in all environments. This issues is resolved. Secure audio and video transfer is now possible in all systems.
- Improvement (SDP): The system now offers automatic SDP error correction, to support (and fix if possible) communication streams initiated with a broken SDP request. This fix improves connectivity to video camera streams.
- Improvement (Compatibility): We made the SDK compatible with older Visual Studio versions, such as Visual Studio 2012.
- Improvement (Dependendency): The system became independent of nuget packages.
- Improvement (GSM/PSTN): SIP trunk access has been improved. Multiple SIP trunk residing on the same IP address, but different ports are now supported. In previous versions only one sip trunk per ip address was allowed.
- Improvement (SIP softphone): SIP softphone registration is improved, to support simultaneous connections with SIP trunks on the same link.
- Improvement (Code review): This version has many bug fixes, and optimizations. The code was optimized to achieve the same functionality with fewer lines of code. Code duplications were removed.
- Improvement (Android compatibility): This version is now fully compatible with the Ozeki SDK for Android. Thes same code will work on both Android mobile phones and Windows and Linux PC-s. Check out https://android-voip-sip.com/ for the Android version
- Bug fix (SRTP): The secure RTP transmission did not work properly in previous versions, now SRTP works well
- Bug fix (Install upgrade): The installer upgrade feature is fixed
- Bug fix (My first phone): The redial functionality is fixed
- Bug fix (Warnings): The reason for many compiler warnings are removed
- Improvement (Version numbers): The version numbering scheme is aligned to the new Ozeki Version Numbers, that is why there is a jump in the version
- Improvement (Android): Preparations were made to make the SDK ready for Android Mobile phones
- Bug fix (Memory leaks): Memory leaks were discovered and fixed in this version related to unsuccessful VoIP calls
- Improvement (Memory consumption): This version is dedicated to memory optimization. Several minor memory leaks are fixed, that were reported by our customers or found by us. The overall memory usage has been reviewed and optimizations were made where we fought it was neccessary. The video frame processing in the computer vision modules could also result in increased memory consumption. This issue was also addressed.
- New feature (Camera / Browser camera supprot): Browser camera support was developed to support HTML5 access to the camera
- Improvement (Camera / MJPEG Streaming): The MJPEG streaming capabilities has been improved greatly. Now it allows switching resolution during streaming, and the resource usage is reduced.
- Improvement (VoIP / relayed calls): Voice calls work well even if all parties are behind firewalls. Now the caller, the callee, and the pbx can operate behind NAT using private IP addresses. In the previous versions the relay calls did not work reliably if all parties were behind firewalls, because some SIP clients (e.g. Bria softphone on Android) were not able to provide appropriate IP address information. This situation is no fixed, the PBX can determines its own public ip using STUN, or this address can be entered manually on the config form, and the PBX can determine the IP addresses of the SIP clients if neccessary using the source adddress of the incoming packets instead of the often unreliable information provided in the SDP data.
- Improvement (Alphanumeric SIP addressing): The PBX can operate using alphanumeric addresses instead of simply numbers. You can setup extensions using phone number and/or names if you develop a PBX
- Improvement (Alphanumeric SIP dial plans): If you build a VoIP PBX, you may setup call routing using alphanumeric addressing.
- Bug fixes: several bugs have been fixed, the list would be long to detail
- Improvement (Contact ID): The contact ID protocol is used in alarm systems. Alarm system clients dial a central server and report alarms as they happen through the phone line. In this version both the server side (Contact ID receiver) and the client side (Contact ID sender) of the contact ID protocol has been improved. Handshake detection and timing has became more precise, and more fault tolerant. Repeat transmissions are handled. Failed and partially failed transmissions and other transmission related error detection is handled in a much better way in both sides. Tests have been performed with multiple base stations.
- Improvement (Onvif): The OnVif camera compatibility has been improved. Some OnVif cameras require additional discovery messages beforre a connect message, in order to connect properly. This issue has been addressed in this version.
- Optimization (Preparing for Android support): This version had many changes under the hood. We are bringing a lot of functionality to an abstract level, to prepare the SDK to be used on Android mobile phones. You might notice some small changes in the namespaces. We eliminated some to make things more simple
- Removed feature (Adobe Flash / media gateway): We have removed the Adobe Flash media gateway support from this version to make the ozekisdk.dll smaller and the code more efficient. Only a very small number of customers used flash Audio streaming. If you need this functionality, send us an e-mail with your license code, and we will help you.
- Improvement (Examples): The examples that come with the SDK have been reviewed and updated where we could make it easier to understand.
- This version does not contain any new features. This version was dedicated to code optimization, cleanup and performance improvement. You will notice that the size of the ozekisdk.dll became smaller, and the number of calls and the complexity of the calls to achieve the desired functionality is optimized.
- Bug fix: Local ports free up after unregistering Ozeki Softphone from Ozeki PBX.
- There was bad sound quality in the Conference Room when the connected VoIP phones sent the RTP packets in different intervals. It is fixed now.
- Bugfix in receiving RTP packets
- Improvements: Snapshot handler has been improved on Linux and Arm platforms.
- Bugfix in network interface handling
- Bugfix in Mpeg4Recorder class
- Improvement in Linux and Raspberry Pi versions. Both platforms have the same installer from this SDK version.
- Improvement in audio quality handling.
- Improvement in Mpeg4Recorder class. Viewing live camera stream will have no delay during recording the stream.
- Improvement in connecting to Mjpeg streams.
- Bugfix in keepalive, wav recording and audio playing (mp3 and wav) functions in the Ozeki Demo Softphone (00_OzekiDemoSoftphoneWPF).
- Bugfix in SnapshotHandler class.
- Bugfix in GrayScale filter.
- Bugfix in PhoneCallAudioReceiver class. In some cases there was an exception when a PhoneCallAudioReceiver object was connected to another MediaHandler. It is fixed now.
- Examples have been updated.
- New Feature: It is possible to read Bluetooth Headset buttons with Ozeki VoIP SIP SDK. On this page you can find a detailed tutorial for it: Link
- Bugfix in attended transfer function
- Bugfix in RTP stream.
- User-Agent configuration has been simplified for SoftPhones and PBX-es.
- CameraServer performance improvement for streaming HD content.
- Bugfix in ContactIdHandler.
- Improvement in ContactIdHandler. Now it completely supports the Ademco ContactID Protocol specification (SIA DC-05-1999.09)
- Bugfix in SDP parser
- Bugfix: The FrameCapture class is no longer capable to read picture files from a folder and send them as frames. We implemented this specific feature in a new class: FrameCaptureFromFolder and we also fixed well known bugs in the new implementation.
- Improvment in RTSP Camera connection.
- Improvement in SIP registration on PBX side. Some SIP clients were not able to connect to Ozeki PBX. Those clients are compatible from this version.
- Bugfix in call handling on PBX side. Blind transfer function did not work when multiple transfers were made. It is fixed.
- Bugfix in playing audio files on Linux and ARM based operating systems.
- Bugfix in call handling related to memory leak issue.
- Bugfix in IP camera connection handling. In some cases it did not work when you disconnected from an IP camera and then reconnected to the camera. It is fixed now.
- Bugfix in Linux SDK. Exceptions on Linux fixed.
- Improvement in the sound quality of PCMU, PCMA and G.722 codecs.
- Improvement in Answer Machine Detector: It detects more efficiently whether the other party is a machine or human.
- Improvement in G.722 codec quality.
- Improvement in the sound quality of Google Text To Speech.
- New Feature: In case of providing an RTSP URL in the Camera Url Builder that contains the username and the password, Ozeki SDK can parse the data from the URL, so you do not have to provide the username and the password again.
- New Feature: It is possible to change the background color of a TextOverlay object.
- Bugfix: Connections between some AudioHandlers were wrong, which caused bad sound quality. This has been fixed.
- New Feature: Google Speech API support. From this version the TTS and the STT functions of Google can be used to convert text to speech and speech to text using the available languages of Google Platform.
- Improvment in sample applications.
- Improvment in RTSP Camera connection.
- Improvment in ONVIF connection: UDP and TCP changing now works flawlessly.
- Bug fix in Unhandled exceptions.
- Improvment in phoneline registration to PBXs: The issue when the same phone numbers are registered to two PBXs and the calls are received by the wrong one is now fixed.
- Improvment in AudioQualityEnhancer: AudioQualityEnhancer has a Start and Stop function. It sends through the data unprocessed without Starting.
- Network checking before sending REGISTER.
- SUBSCRIBE message handling in PBX.
- Improvment in video calls: The disconnection issue is now fixed.
- New feature: Opus codec is available from this version. You can make and receive VoIP calls using Opus codec.
- Bugfix in SIP message handling.
- Bugfix in USB Camera handling.
- Bugfix in connecting to Mjpeg streams.
- Improvement in VideoViewer class: It is possible to disable the context menu from this version.
- Bugfix in sending Contact ID messages.
- Bugfix in managing camera stream profiles.
- Bugfix in handling resources.
- Improvement in RTSP handling.
- Improvement in connecting to mjpeg streams.
- Bugfix in the Attended transfer function.
- Bugfix in the UDP TransportType of the DirectIpPhoneLine.
- Bugfix in escaping characters in SIP headers.
- Bugfix in webcamera recorder. Now it is possible to record videos longer than 30 minutes.
- Bugfixed memory leak in ComputerVision examples.
- Bugfix at moving calls between different ConferenceRoom objects.
- Improvement in h264 video handling. Now h264 videos are faster and have better quality.
- This version is able to connect to SIP servers which have multiple IP addresses.
- No more exceptions occur in camera connections and SIP messaging.
- Bugfix in streamer examples. It is able to connect to all streamer examples via RTSP.
- New feature: Color tracking function (read more here).
- Bugfix in ContactIdHandler. It is able to send multiple contact id messages.
- Bugfix in IncomingCall event. From this version it will not occur multiple times.
- The class DeviceInfo in the Ozeki.Media namespace has been renamed to AudioDeviceInfo to better describe its role and to avoid ambiguity.
- New feature:Native HTML5 video streaming
- New feature:.deb packages for the experimental Linux/Mono version of the SDK:
Ozeki_SDK_1.6.1.deb - only compatible with PC
Ozeki_SDK_1.6.1_armhf.deb - only compatible with Raspberry Pi
- New feature: WebMStreamer class. From this version it is possible to stream video and audio together to a website using the video tag of HTML5. See: 14_WebMStream_For_HTML5 example (\Ozeki\Ozeki SDK\Examples\Camera\05_Advanced\14_WebMStream_For_HTML5).
- Improvement: VideoViewer turns black when the camera connection ends.
- Improvement: It is possible to provide the number of extensions which you wish to subscribe to (presence event):
var subscription = _phoneLine.Subscription.Create("presence", "100"); subscription.NotificationReceived += subscription_NotificationReceived; _phoneLine.Subscription.Subscribe(subscription);
- Bugfix: You can get the type of incoming call if the IncomingCall event occurs (e.Item.CallType).
- Bugfix: System.InvalidOperationException of System.Drawing.Image.Clone() in Ozeki.Media.VideoViewerWF has been fixed.
- Bugfix: CPU issue has been fixed in case of using the TCP transport type.
- New feature: SDP information can be saved during RTSP connection. See: BasicIPCameraViewer example.
- Bugfix in RTSP handling.
- Bugfix in Contact ID Handler: Multiple Contact ID messages can be sent and the number of sending attempts can be queried.
- Bugfix in Mpeg4Recorder class.
- Bugfix in Linux dll
- Bugfix in MJPEGConnection class
- Bugfix in MJPEGStreamer class
- Bugfix in CreateDirectIPCallObject method
- Bugfix in RTSP handling
- Bugfix in Mpeg4 recorder
- Minor bugfixes
- Improvement in MjpegStreamer: From this version Ozeki SDK calculates the number of frames that are sent through the network based on the bandwidth. This means that the packets are not lost on the network, and the latency becomes minimal. You can find an example here.
- Ozeki SDK for Linux: Ozeki SDK has become available on Ubuntu based Linux distributions from this version. You can develop Camera viewer, streamer and VoIP applications using Mono Framework. You can read more about the Linux version here.
- Bugfix in camera connection
- Minor bugfixes
- Bugfix in H.264 codec
- Bugfix in RTP packet handling
- Bugfix in RTSP connection handling
- Bugfix in call handling (PBX mode)
- Improvement: It is possible to disable DNS SRV from now.
- Bugfix in DNS resolution.
- Improvement: VideoViewerWPF mouse events
- Improvement in DNS SRV handling
- Bugfix in MjpegConnection class
- Bugfix in PTZ
- Minor bugfixes
- Bugfix in DNS SRV handling
- Bugfix in SIP message handling
- Bugfix in Ozeki Demo Softphone
- Bugfix in RTSP
- Bugfix in DemoSoftphone
- Bugfix in Call State Change in PBX Force Relay mode
- Bugfix in Outbound Call mapping
- Bitmap converter improvement
- AudioQualityEnhancer improvement
- Bugfix: BigConnect softphone and Cisco Jabber Video Client are now compatible.
- Bugfix in DNS SRV resolv.
- Bugfix in GainControl class.
- New feature: OggReader class and OggStreamPlayback class.
- New feature: DNS SRV is now supported.
- Bugfix in memory handling.
- New feature: Incoming and outgoing RTP data can be accessed.
- Bugfix in network interface handling.
- Bugfix in Answering Machine Detector.
- New feature: VideoViewerWF Mouse events
- New feature: RTCP sender report
- Bugfix in ONVIF listener
- Bugfix in Camera Server
- Bugfix in Mp3StreamRecorder class.
- Bugfix: Now we can play mpeg4 videos with AVPlayer class recorded by Mpeg4Recorder class.
- Bugfix in MjpegStreamer class.
- Bugfix in BitMapConverter class.
- RTSP handling improvement
- Bugfix: Now we can play our RTSP stream.
- Bugfix in MjpegStreamer class.
- Bugfix in CameraUrlBuilder classes.
- Bugfix in examples and demo applications.
- Asserts have been removed.
- Bugfix in Mp3StreamRecorder class
- Bugfix in BitmapConverter class
- Bugfix in CreateCallObject method
- Bugfix in Ozeki IP Camera Manager Demo
- Bugfix in AudioQualityEnhancer
- H264 Decoder improvement
- RTSP handling improvement
- Bugfix in Ozeki Demo Softphone
- Bugfix in Ozeki IP Camera Manager Demo
- Bugfix: Sometimes receiving peer to peer calls did not work.
- Bugfix: There was a space character in the SDP in case of TLS transport type. It has been removed.
- Bugfix: SDK did not send back ACK SIP message when call type was modified. It works now.
- Bugfix: Tripwire event sender has been fixed.
- Improvement: VideoConcat class
- Improvement in MjpegStreamer
- Bugfix in Mpeg4Recorder
- Bugfix in H264 encoder and decoder
- Namespace correction
- New feature: New MediaHandler (VideoConcat - it is able to concatenate multiple video sources)
- New example: VideoConcat (\Ozeki\Ozeki SDK\Examples\Camera\05_Advanced\13_VideoConcat)
- Bugfix: Bugfix in Ozeki IP Camera Manager and Ozeki Demo Softphone applications
- RTSP handling improvement
- ONVIF handling improvement
- New feature: Tampering
- New example: Tampering detector (Camera\5_Advanced\11_Tampering)
- Minor bugfixes
- TextOverlay improvement
- Mpeg4Recorder improvement (e.g. FramesCount property)
- Documentation bugfix
- New MediaHandler: VideoCodecConverter (you can modify the video format from raw format to h263\h263+\h264 formats)
- ImageMask improvement (selectable mask color)
- DeviceDiscovery bug and memory leak fix
- Instant message response received event raised
- New Datareceived event in the WaveStreamRecorder to catch the raw data
- AudioQualityEnhancer bugfix
- New examples: Camera\5_Advanced\10_Image_Mask_WF
- VideoViewer improvement
- WPF controls bugfix
- Namespace simplification
- Examples correction and improvements (e.g. Ozeki Demo Softphone)
- Native exception handling
- Bugfix in .NET 3.5 dll (VoIP)
- A bug was discovered in speaker handling. It is fixed.
Ozeki SDK is the next generation of our VoIP SDK library. This
is an amazing new development with huge improvements in the field
of camera handling. We have simplified the namespaces, optimized the
code and fixed several annoying bugs. Please download, and install
this version, and check the examples folder to see what you can do
with this fantastic new tool.
Some of the improvements:
- The number of namespaces have been reduced
- Detailed network communication logging is available to trace your calls
- Memory management has been optimized. Significantly less memory is used for video and audio calls.
- Performance has been optimized. Less CPU is used for audio and video encoding and decoding
- Bug fixes in codec implementations, such as H264, AAC, etc
- Improved RTSP streaming (audio/video)
- IP camera support with Pan/Tilt/Zoom (PTZ) capability
- OnVif protocol support (www.onvif.org)
- MJPEG streaming added to be able to stream video calls to webbrowsers without Flash, Silverlight or WebRTC support
- Computer vision capabilities can be used in SIP video calls (Number plate recognition, Face detection, Line detection, Circle detection, etc...)
- Optimized multi way audio and video streaming can be set up to be able to broadcast voice and video
- Device discovery was added for local networks
- There are several other improvements, please check the examples folder.
- Example improvements
- Minor bugfixes
- Example improvements
- Minor VoIP and IPCamera bugfixes
- NuGet example update
- IPCamera ConnectionLost timeout can be set on API
- RTSP handling improvement
- Instant Message sending without being in InCall state
- IPCamera RTSPClient refactor
- IPCamera RTP over RTSP (TCP interleave) support
- Minor bugfixes
- CultureInfo for grammar can be set on SpeechToText
- NVA related bugfixes
- New feature: License Plate Recognizer
- New feature: Tripwire
- Improvement in Answering Machine Detection
- NVA example applications included
- Minor bugfixes
- New library included: Network Video Analytics
- line detection
- edge detection
- corner detection
- circle detection
- barcode scanner
- New feature: Laser Distance Measurement
- Silverlight 5 library included for Media Gateway Client SDK
- API improvements (ISpeaker, IMicrophone, IMediaConnector interfaces)
- Minor bugfixes
- Parameters can be changed for Waveform speaker
- IP camera connection improvements
- ONVIF snapshot feature without connecting to camera device
- IP range can be specified when discovering ONVIF devices
- Minor bugfixes
- Selectable video resolution during calls
- ONVIF IP Camera Manager application improvement
- ONVIF IP Camera PTZ improvement
- ONVIF IP Camera Networking improvement
- ONVIF IP Camera static ONVIF snapshot command
- Minor bugfixes
- Custom source identifier can be set on IPhoneLine
- ONVIF IP Camera Manager application improvement
- Minor bugfixes
- IP camera connectivity support
- New installer
- Launcher application included
- Minor bugfixes
- Fixed some issue with API class visibility
- Fixed some issue with video encoding
- Fixed the issue with VOS2000 registration
- Media features and improvements:
- Better performance
- No audio mixer required when connecting multiple audio sources to a destination
- Different audio can be played in the left and right channel simultaneously
- Improvement in audio and video synchronization
- Audio/Video handlers can support multiple audio/video formats
- SIP features and improvements:
- Extended registration info for IPhoneLine and SIP extensions
- Multiple SIP account registration with the same port number
- Subscription to any SIP event-packages via SUBSCRIBE requests
- SIP header mapping for SIP extensions and IPhoneLine
- Trusted network identity support (P-Asserted-Identity, P-Preferred-Identity, Remote-Party-ID)
- Notification about MESSAGE sending failure
- Call features and improvements:
- Easier call parameter customization (such as caller ID or call type)
- SRTP mode can be specified for each calls
- Optimized direct IP P2P calls
- Custom properties can be used in ICall objects
- PBX features and improvements:
- Simplified extension identification
- Simpler call handling for extensions (no need to create it manually via IPBXCallFactory)
- Much more easier to implement SIP trunk and VoIP provider connection
- Revised dial plan parameters (easier to get caller ID and dialed number information)
- SessionMode can be specified for each calls in the dial plan
- New sample projects
- Several API modification
- Improvement in TCP stream handling
- New feature: Contact Id protocol support
- The aggregated H264 NAL unit is supported in RTP stream
- Registration issue is fixed in Asterisk based PBXes
- SIP register compatibility with VOS2000/3000 PBX
- Answering machine detector related bugfixes
- SIP reason phrase can be accessed when error occurred during a call
- Improved compatibility with SIP devices without 'rport' support
- Improved video recording with H.263 codec
- Minor SIP related bugfixes
- Incoming RTP data can be accessed in audio and video handlers
- Improvement in VAD filter
- UTF-8 character encoding support in SIP messages
- Improvement in dial plan: new destination types introduced (reject, redirection)
- Minor bugfixes
- New feature: Video recording
- Improvement in video and audio synchronizing
- Fixed SIP port can be specified when creating the IPhoneLine
- PBX relay mode can be set when routing the call
- Minor bugfixes
- Improvement in speech-to-text
- Custom speech-to-text engine implementation can be used
- Word recognition mode can be specified
- An improved version of the default demo softphone is included
- New feature: TLS support in PBX module
- Improved compatibility with firewalls and routers
- Minor bugfixes
- Real-time video quality change
- Improvement in call media handling
- Simplified codec selection
- SIP identity can be set on a direct IP phone line
- Revised logging system
- Minor bugfixes
- Improved codecs
- Improvement in PBX authentication
- An improved version of the speaker can be selected
- Supported OS: Vista and above
- Automatic DTMF signalling mode detection during a call
- The TextToSpeech MediaHandler can be extended with other TTS engines
- Modifications on IPhoneLine interface
- More information available about the SIP registration at client side (IPhoneLine.RegistrationInfo)
- RegisteredInfo renamed to LineState
- Modifications on ICall interface:
- New property: IsAnswered, indicates whether the call has been answered
- New method: HoldCall() for putting the call on hold
- New method: UnholdCall() for taking off the call from hold
- The older Hold() method has been renamed to ToggleHold()
- New CallState has been added: Answered
- See the sample projects for optimal usage
- Improvement in custom SIP/SDP message modification
- Improvement in codec selection
- The availability of the WebCamera can be checked with the Initialized property
- Minor bugfixes
- New feature: Making and receiving P2P calls without SIP server
- New feature: Multipart SIP message body support
- New feature: Static public IP can be set in NAT configuration
- New feature: Handling received blind transfer (REFER) requests at client side
- Direct IP session setup for VoIP phones
- Peer-to-peer SIP session setup
- Improvement in transport error management
- New properties on IPhoneLine interface: LocalAddress, LocalPort
- Improvement in PBX call routing
- Improvement in PBX messaging
- Minor compatibility fixes
- New feature: video bitrate can be set.
- Improvement: it is possible to easily and simply define the type of codecs you wish to use.
- PBX improvement: more efficient caller ID and dialed number handling.
- Minor bugfixes.
- New feature (New CallState): InactiveHeld. This call state occurs when both parties place the call on hold
- Improvement (SIP account): more flexible domain validation during specifying SIP account
- Improvement (PBX module): SIP authentication related improvements in the PBX module.
- Improvement: optimized Ozeki DemoSoftphone WPF
- Minor bugfixes
- Improvement: TLS authentication has been fixed
- Improvement: more stable calls in the PBX module (ISession and CallManager have been changed)
- New feature: it is possible to customize speech rate in TextToSpeech
- New feature: a MediaHandler that can play unique formats can be derived from AudioStreamPlayback class
- New feature: it is possible to select the type of voice that will be played when the maximum number of calls is reached in the PBX module of the licensed Ozeki VoIP SIP SDK
- SIP TCP connection improved
- MediaGateway FileNotFoundException fixed
- Improved transfer compatibility with Cisco and Avaya
- New feature: You can send DTMF signal over SIP info.
- BUG fix: SIP register compatible with SipXecs PBX.
- BUG fix: WP7 packet handling improved.
- New feature: You can create Windows Phone VoIP application. Many examples included.
- New feature: PBX can handle multi-network interface.
- Bug: Minor bug fixes in PBX phone calls.
- New feature: NAT discovery.
- Bug fix: microphone instance freezes when microphone device is disconnected.
- Reduced installation time.
- Trial time calculation fixed.
- SetKeepalive method was removed from ISoftPhone.
- VoIP SDK is compatible with voipgateway.org.
- DTMF event player has changed. You can play dtmf sound anytime(call is not required).
- Trial version limitation reduced: You can create unlimited simultaneous phone calls and phone lines.
- Incoming calls will not be rejected because of the licensing.
- New feature: You can add custom phone number, display name in the phone call.
- Improvement in LicenseManager.
- Improvement in call forwarding.
- New feature: TLS support.
- New feature: attended transfer.
- Improvement in Avaya connectivity.
- New feature: SRTP.
- New feature: blind transfer.
- New video codec: H.263.
- New audio codecs:
- L16
- G723
- G726-16
- G726-24
- G726-32
- G726-40
- G728
- Improvement in echo cancellation.
- Webphone development support (Flash and Silverlight).
- Improvement in silence filter.
- Minor bug fixes.
- New feature: Answering machine detection.
- Improvement in Voice Activity Detection.
- Minor bug fixes.
- New feature: Video calls are supported effectively from this version.Available video codecs:
- H.263+
- H.264
More information