Ozeki VoIP SDK - Product Guide
Did you know?
This SDK was used to build:Ozeki Phone System XE - VoIP PBX Software for Developers Which is a high performance PBX system supporting Mobile and Desktop phones.
It was also used to create Ozeki 3D VoIP softphone. A cool SIP client that allows 3D Video calls.
Achieve extended codec support with Ozeki VoIP SIP SDK
This page enlists the supported codecs by Ozeki VoIP SIP SDK. By clicking on the links below you can find additional information about the selected codec. Therefore you can get some knowledge about it as a user to be able to make the right decision relating codecs.
A codec is a device or computer program that is for encoding and/or decoding digital data stream or signals. In VoIP technology codecs help convert analog voice signals to digitally encoded version. There is a wide range of different codec types as codecs can be vary according to their bandwidth, sound quality or computational requirements, etc.
Codecs encode data streams or signals to be able to transmit, store or encrypt them, or decode signals for playback or editing. Codecs are used for videoconferencing and streaming media applications.
Audio compressors allow to convert analog audio signals into digital ones to transmit or store them. Then the receiving device can convert the digital signals back to analog ones with the use of an audio decompressor, for playback.
Lossy codecs: Lossy refers to the fact that compression results in some quality reduction. Algorithms are also used to create the impression of data being there.
Lossless codecs: These types of codecs are used to archive data in a compressed form in a way that all information of the original stream is kept. They are the proper choices if it is more important to retain the original quality than reduce data sizes (especially in cases when the data is undergo further processes).
General Information
Codec name | Creator | Implementations (codecs) | Application | Patented |
---|---|---|---|---|
G.711
Learn more |
ITU-T | various VoIP software | voice recording, telephony | No |
G.722
Learn more |
ITU-T | various VoIP software | voice recording, telephony | Yes |
G.729
Learn more |
ITU-T | various VoIP software | voice recording, telephony | Yes |
iLBC
Learn more |
Global IP Solutions | various VoIP software, Cisco IP Communicator[15], old versions of Skype | voice recording, telephony | ? |
Speex
Learn more |
Xiph.Org Foundation | Flash Player | voice recording, telephony | No |
Technical details
Codec name | Algorithm | Sample Rate | Bit rate | Bits per sample | Latency | CBR | VBR | Stereo | Multi - channel |
---|---|---|---|---|---|---|---|---|---|
G.711 Learn more |
companding A-law or μ-law, PCM, Lossy | 8 kHz | 64 kbit/s | 13 bit | 125ms | Yes | No | No | No |
G.722 Learn more |
Lossy | 16 kHz | 24, 32 kbit/s | 16 bit | 40 ms | Yes | No | No | No |
G.729 Learn more |
CS-ACELP, Lossy | 8 kHz | 8 kbit/s | 13 bit | 15 ms | Yes | No | No | No |
iLBC Learn more |
Lossy | 8 kHz | 13.33, 15.20 kbit/s | 16 bit | 30, 20ms | Yes | No | No | No |
Speex Learn more |
Lossy, Speech | 8, 16, 32 (48) kHz | 2.15 to 24.6 kbit/s (NB); 4 to 44.2 kbit/s (WB) | 16 bit | 30ms (NB) 34ms (WB) | Yes | Yes | Yes | Yes |
GSM-HR Learn more |
Lossy | 8 kHz | 5.6 kbit/s | 13 | 25ms | Yes | No | No | No |
GSM-FR Learn more |
Lossy | 8 kHz | 13 kbit/s | 13 | 20-30ms | Yes | No | No | No |
GSM-EFR Learn more |
ACELP, Lossy | 8 kHz | 12.2 kbit/s | 13 | 20-30ms | Yes | No | No | No |
BEGINNER
Getting started
Downloading VoIP SIP SDK
Installation steps
PBX configuration
Examples with source code
INTERMEDIATE
VoIP technology walkthrough
SIP softphone development
Webphone development
Mobile development
Voice recording
GETTING AROUND
Sitemap
Search the manual
API documentation
FAQ
Acknowledgements