ILBC
Encyclopedia
Internet Low Bitrate Codec (iLBC) is an open source
royalty-free narrowband
speech codec
, developed by Global IP Solutions
(GIPS) formerly Global IP Sound (acquired by Google Inc in 2011). It was formerly licensed as a freeware
with limited commercial use, but since 2011 it is available under an open source (3-clause BSD license) license as a part of the open source WebRTC
project. It is suitable for VoIP applications, streaming audio, archival and messaging. The algorithm is a version of block-independent linear predictive coding
, with the choice of data frame
lengths of 20 and 30 milliseconds. The encoded blocks have to be encapsulated in a suitable protocol for transport, usually the Real-time Transport Protocol
(RTP).
iLBC handles the case of lost frames through graceful speech quality degradation. Lost frames often occur in connection with lost or delayed IP packets. Ordinary low-bitrate codecs exploit dependencies between speech frames, which unfortunately results in error propagation when packets are lost or delayed. In contrast, iLBC-encoded speech frames are independent and so this problem will not occur.
iLBC is defined in RFC 3951. It is one of the codecs used by Gizmo5
, webRTC
, Ekiga
, QuteCom, Google Talk
, Yahoo! Messenger
, Polycom IP Phone and Maemo Recorder (on the Nokia N800
/N810
) and many others.
iLBC was submitted to IETF in 2002 and the final specification was published in 2004.
Open source
The term open source describes practices in production and development that promote access to the end product's source materials. Some consider open source a philosophy, others consider it a pragmatic methodology...
royalty-free narrowband
Narrowband
In radio, narrowband describes a channel in which the bandwidth of the message does not significantly exceed the channel's coherence bandwidth. It is a common misconception that narrowband refers to a channel which occupies only a "small" amount of space on the radio spectrum.The opposite of...
speech codec
Codec
A codec is a device or computer program capable of encoding or decoding a digital data stream or signal. The word codec is a portmanteau of "compressor-decompressor" or, more commonly, "coder-decoder"...
, developed by Global IP Solutions
Global IP Solutions
Global IP Solutions is a United States-based corporation that develops real-time voice and video processing software for IP networks. The company delivers embedded solutions that enable real-time communications capabilities for video and Voice over IP...
(GIPS) formerly Global IP Sound (acquired by Google Inc in 2011). It was formerly licensed as a freeware
Freeware
Freeware is computer software that is available for use at no cost or for an optional fee, but usually with one or more restricted usage rights. Freeware is in contrast to commercial software, which is typically sold for profit, but might be distributed for a business or commercial purpose in the...
with limited commercial use, but since 2011 it is available under an open source (3-clause BSD license) license as a part of the open source WebRTC
WebRTC
WebRTC is a framework that was open sourced on June 1, 2011 that allows web browsers to conduct real-time video chat.Its inclusion in the World Wide Web Consortium standards is supported by Google, Mozilla and Opera....
project. It is suitable for VoIP applications, streaming audio, archival and messaging. The algorithm is a version of block-independent linear predictive coding
Linear predictive coding
Linear predictive coding is a tool used mostly in audio signal processing and speech processing for representing the spectral envelope of a digital signal of speech in compressed form, using the information of a linear predictive model...
, with the choice of data frame
Data frame
In computer networking and telecommunication, a frame is a digital data transmission unit or data packet that includes frame synchronization, i.e. a sequence of bits or symbols making it possible for the receiver to detect the beginning and end of the packet in the stream of symbols or bits...
lengths of 20 and 30 milliseconds. The encoded blocks have to be encapsulated in a suitable protocol for transport, usually the Real-time Transport Protocol
Real-time Transport Protocol
The Real-time Transport Protocol defines a standardized packet format for delivering audio and video over IP networks. RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, television services and...
(RTP).
iLBC handles the case of lost frames through graceful speech quality degradation. Lost frames often occur in connection with lost or delayed IP packets. Ordinary low-bitrate codecs exploit dependencies between speech frames, which unfortunately results in error propagation when packets are lost or delayed. In contrast, iLBC-encoded speech frames are independent and so this problem will not occur.
iLBC is defined in RFC 3951. It is one of the codecs used by Gizmo5
Gizmo5
Gizmo5 was a Voice over Internet Protocol communications network and a proprietary freeware soft phone for that network. On November 12, 2009, Google announced that it had acquired Gizmo5...
, webRTC
WebRTC
WebRTC is a framework that was open sourced on June 1, 2011 that allows web browsers to conduct real-time video chat.Its inclusion in the World Wide Web Consortium standards is supported by Google, Mozilla and Opera....
, Ekiga
Ekiga
Ekiga /i k ai g a/ is a VoIP and video conferencing application for GNOME and Windows. It is distributed as free software under the terms of the GNU General Public License. It was the default VoIP client in Ubuntu until October 2009, when it was replaced by Empathy...
, QuteCom, Google Talk
Google Talk
Google Talk is a freeware voice over Internet protocol client application offered by Google Inc. The first beta version of the program was released on August 24, 2005...
, Yahoo! Messenger
Yahoo! Messenger
Yahoo! Messenger is an advertisement-supported instant messaging client and associated protocol provided by Yahoo!...
, Polycom IP Phone and Maemo Recorder (on the Nokia N800
Nokia N800
The Nokia N800 Internet tablet is a wireless Internet appliance from Nokia, originally announced at the Las Vegas CES 2007 Summit in January 2007. N800 allows the user to browse the Internet and communicate using Wi-Fi networks or with mobile phone via Bluetooth. The N800 was developed as the...
/N810
Nokia N810
The Nokia N810 Internet tablet is an Internet appliance from Nokia, announced on 17 October, 2007 at the Web 2.0 Summit in San Francisco. Despite Nokia's strong association with cellular products, the N810 is not a phone, but instead allows the user to browse the Internet and communicate using...
) and many others.
iLBC was submitted to IETF in 2002 and the final specification was published in 2004.
Parameters and features
- Sampling frequency 8 kHz/16 bit (160 samples for 20 ms frames, 240 samples for 30 ms frames)
- Controlled response to packet lossPacket lossPacket loss occurs when one or more packets of data travelling across a computer network fail to reach their destination. Packet loss is distinguished as one of the three main error types encountered in digital communications; the other two being bit error and spurious packets caused due to noise.-...
, delay and jitterJitterJitter is the undesired deviation from true periodicity of an assumed periodic signal in electronics and telecommunications, often in relation to a reference clock source. Jitter may be observed in characteristics such as the frequency of successive pulses, the signal amplitude, or phase of... - Fixed bitrateBitrateIn telecommunications and computing, bit rate is the number of bits that are conveyed or processed per unit of time....
(15.2 kbit/s for 20 ms frames, 13.33 kbit/s for 30 ms frames) - Fixed frame size (304 bits per block for 20 ms frames, 400 bits per block for 30 ms frames)
- Robustness similar to pulse code modulation (PCM) with packet loss concealmentPacket Loss ConcealmentPacket loss concealment is a technique to mask the effects of packet loss in VoIP communications. Because the voice signal is sent as packets on a VoIP network, they may travel different routes to get to destination. At the receiver a packet might arrive very late, corrupted or simply might not...
, like the ITUInternational Telecommunication UnionThe International Telecommunication Union is the specialized agency of the United Nations which is responsible for information and communication technologies...
-T G.711G.711G.711 is an ITU-T standard for audio companding. It is primarily used in telephony. The standard was released for usage in 1972. Its formal name is Pulse code modulation of voice frequencies. It is required standard in many technologies, for example in H.320 and H.323 specifications. It can also... - CPU load similar to G.729A, with higher basic quality and better response to packet loss
- Royalty-free
- Commercial use of the source code supplied by GIPS requires a license
- PSQMPSQMPSQM is a computational and modeling algorithm defined in ITU Recommendation ITU-T P.861 that objectively evaluates and quantifies voice quality of voice-band speech codecs....
testing under ideal conditions yields mean opinion scoreMean Opinion ScoreThe Mean Opinion Score test has been used for decades in telephony networks to obtain the human user's view of the quality of the network. In multimedia especially when codecs are used to compress the bandwidth requirement , the mean opinion score ...
s of 4.14 for iLBC (15.2 kbit/s), compared to 4.3 for G.711G.711G.711 is an ITU-T standard for audio companding. It is primarily used in telephony. The standard was released for usage in 1972. Its formal name is Pulse code modulation of voice frequencies. It is required standard in many technologies, for example in H.320 and H.323 specifications. It can also...
(µ-law)