| Ozeki VoIP SIP SDK |
Product information |
| Product name |
Ozeki VoIP SIP SDK |
| Category |
Software Development Kit |
| Product website |
https://www.voip-sip-sdk.com |
| Latest version | Ozeki SDK 10.4.13 |
|---|
| Release date | 2018.03.08. |
|---|
| Package size | 138.3 MB |
|---|
| Download url | https://www.voip-sip-sdk.com/p_7021-download-ozeki-voip-sip-sdk.html |
|---|
| Package contents |
- Redistributable .DLL
- Documentation
- Example applications
- Exe demo
- Full source code (optional)
|
| Main task |
Makes it possible to build a VoIP client application or a PBX, based on the SIP protocol. |
| Connectivity |
It connects to a supported VoIP PBX or
to a VoIP service provider over the Internet. Supports firewall passthrough (STUN/TURN). |
| Supported client OS |
- Windows server 2022
- Windows server 2019
- Windows server 2016
- Windows 8
- Windows 10
- Windows 11
|
Required .NET
framework |
At least .NET Framework 4.5.2 or any newer version |
Supported
programming
languages and
environments |
Microsoft Visual Studio 2022, 2019, 2017, 2015, (C#, VB.NET, ASP.NET,...)
|
| Source code |
Full source code can be purchased.
The source code of this VoIP SDK is in C#.Net. |
| Developer features |
- Easy to use
- Very easy to incorporate
- Makes quick development possible
- Supports all development environments with .NET support
- Supports the development of WPF, Windows Form, Windows Service application, etc.
- Free product version updates: one year free updates
|
| Call features |
- Multiple simultaneous call support
- Make and receive Audio/Video calls
- Make and receive peer-to-peer calls without a SIP server
- Call reject
- Hold/Unhold
- Call forward
- Call transfer (Attended and Blind transfer)
- DTMF support: send/receive DTMF signals via RFC 2833, SIP INFO or inband
- Do Not Disturb(DND)
- Auto Answer
- Redial
- Secure calls (TLS, SRTP)
- Codec priority change
- Caller ID modification for outbound calls
- Call history
- Multi-party voice conference support (Conference split/join, locally mixed conferences)
|
| SIP features |
- Multiple SIP account support
- Secure SIP connection via TLS
- Message Waiting Indicator (for checking voicemail)
- Send and receive instant messages
- Digest authentication
- Supported SIP methods: REGISTER, INVITE, ACK, CANCEL, OPTIONS, BYE, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
- Direct access to incoming and outgoing SIP messages (add/modify SIP headers for inspect or repair)
- Multipart SIP body handling
- Outbound proxy server support
- Subscription to SIP event packages
|
| PBX features |
- Fully customized dial plan creation (call and message routing)
- Custom extension or custom connection creation (eg. voicemail, SIP trunk, echo test)
- Rich call information and call event notification
- Third-party call control (forward, hold, transfer, hangup)
- Direct access to caller and callee RTP stream outside of a call
- Authentication control
- Music on hold
|
| Network features |
- Multiple network interface support
- Supported protocols: UDP, TCP, TLS, SIP, SDP, RTP, SRTP, STUN, TURN, ICE
- Configurable port range
- Firewall/NAT passthrough (auto discovery, STUN, static IP setting)
|
| Audio features |
- Microphone & Speaker device selection (on-the-fly as well during a conversation/conference)
- Device calibration (volume, level, mute, device change, format change)
- Play wav or mp3 files to remote party
- Record audio in wav or mp3 format
- Text-To-Speech support (changing voice, setting speech rate, multiple TTS engines)
- Speech-To-Text/Speech recognizer support (changing voice, multiple STT engines)
- Play DTMF tones
- Play audio from multiple audio sources to remote party
- HD audio support (HD audio calls)
- Supports most audio formats (8000-48000 Hz, 16bit, mono/stereo)
- Automatic audio format conversion
- Support access incoming and outgoing audio stream directly
- Adaptive jitter buffer
- Packet loss concealment
|
| Video features |
- Camera device selection (on-the-fly as well during a conversation/conference)
- Device calibration (device change, resolution/frame rate change)
- Play video files to remote party (mp4)
- Record video in mp4 format
- 3D video support
- Real-time video quality change
- Picture manipulation (rotate, flip)
- 720p, SVGA, XVGA, VGA, CIF, QCIF video resolutions
- Support access incoming and outgoing video stream directly
|
| Advanced Digital Signal Processor features |
- Auto Gain Control (AGC)
- Noise Reduction
- Voice Activity Detection (VAD)
- Acoustic Echo Cancellation (AEC)
- Answer Machine Detection (predictive dialer)
|
| Supported audio codecs |
- PCMA (G.711 aLaw)
- PCMU (G.711 uLaw)
- GSM
- iLBC (20 and 30 ms)
- Speex Narrowband (8kHz)
- Speex Wideband (16kHz)
- Speex Ultrawideband (32kHz)
- G.722
- G.723
- G.726-16
- G.726-24
- G.726-32
- G.726-40
- G.728
- G.729
- L16
- OPUS
|
| Supported video codecs |
|
| Fields of application |
- Softphones
- Webphones
- Online chat communities (e.g.: dating, business meetings)
- PBX servers
- Automatic call distributors
- Telephone and call centers
- VoIP providers
- Conferencing applications
|
| Supported PBX systems |
|
| Example applications |
|
| Standards |
|
+36 1 371 0150