Polycom Siren/Codecs FAQs


What Is G.722.1?

Approved on September 30th, 1999, after a four-year selection process involving extensive testing, ITU-T Recommendation G.722.1 is a state-of-the-art international standard wideband audio compression algorithm. It is based on Polycom's third-generation Siren compression technology and is derived from Polycom's field-proven PT716plus algorithm. Polycom developed this technology to meet the demanding audio needs of the multimedia community. It provides high-quality audio at low bit rates with low delay and very low complexity. It works for all kinds of audio signal including speech, music, and singing, for example.

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How can I Obtain ITU-T Recommendation G.722.1?

An electronic copy of G.722.1 may be downloaded directly from the ITU at  http://www.itu.int/rec/T-REC-G.722.1-200505-I/en.

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What Exactly Is Contained in ITU-T Recommendation G.722.1?

ITU-T Recommendation G.722.1 consists of the following items:

  • Description of the wideband coding algorithm.
  • Reference C code for the encoder and decoder.
  • Test vectors (signals), a tool to assist implementers in verifying the accuracy of their implementation.

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How Do I Implement G.722.1?

Implementations of audio codecs are available from a variety of vendors. If you wish to create your own implementation, first purchase a copy of G.722.1 from the ITU-T. Recommendation G.722.1 contains all the information required to implement the algorithm. The output signals from any implementation of G.722.1 on any hardware should exactly match that of the reference C code when processing the same identical input signals. The test vectors provided in the standard are designed for the purpose of testing the correctness of an implementation.
There are both input and output test vectors to test the encoder and decoder implementations. These test vectors were created to exercise as much of the algorithm code as possible. Therefore any implementation successfully reproducing the output test vectors is considered to accurately reproduce the reference C code performance.

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Can I get a demonstration of G.722.1?

Polycom has created a demonstration program, SirenZip, that runs on Microsoft Windows 95 or higher versions of the Microsoft Windows operating system. It executes G.722.1 at one of three selectable bit rates (the 16 kbps extension, 24 kbps or 32 kbps). Download your free demonstration copy of SirenZip (download size 208 KB).
How do I use SirenZip once it's downloaded?
Once downloaded, SirenZip is ready to run.

  • Double click on SirenZip.exe, then click on the Siren Encode and enter the correct audio source and output bitstream file names. Select the correct bit rate, then either click on Play Wav Input to hear the input file, or click on Encode Wave Input to compress the file using G.722.1.
  • To decode, click on Siren Decode and enter the correct bitstream source file and audio output file names. Then click on either Play Wav Output to hear the G.722.1 decoded output file, or click on Decode Bitstream to synthesize the audio output using G.722.1.

Here are some points to keep in mind when using SirenZip:

  • The encoder accepts audio input files in a mono wave format.
  • The decoder outputs audio files in a mono wave format.
  • The bit rate may be set in the encoder, the decoder automatically knows the correct operational bit rate.
  • Audio input files longer than 60 seconds will be truncated to 60 seconds.
  • The audio input may be at one of two sample rates, 16,000 Hz or 22,050 Hz. If the input is sampled at 22,050 Hz it will be down sampled to the correct 16 kHz by SirenZip; this may result in some changes to the audio fidelity. It is recommended to use 16 kHz sampling whenever available.

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What are the Technical Specifications for G.722.1?

Complete technical specifications are published in the ITU-T standard for G.722.1. It is specified as a fixed-point algorithm. A floating-point version will be standardized in the future by the ITU-T, and it will interoperate with the fixed-point standard.(An interoperable floating-point version exists at Polycom.) The MIPS complexity numbers below are examples of non-optimized implementations on three different types of DSP. Note, two of the DSPs shown are floating-point units.
General G.722.1 Parameters



Audio sample rate

16 kHz

Bit rate (rate may change on any frame boundary)

24, 32 kbps

Audio Bandwidth

50 Hz - 7 kHz.

Audio frame size

20 ms

Algorithmic delay (see Note 1)

40 ms

RAM (fixed point)

< 7.5 k bytes

ROM table space (fixed point)

~ 20 k bytes

MIPS ratio between encoder and decoder

approximately 1 to 1

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Why Wideband?

Traditional telephony is called "narrowband" because it passes audio signals only in the range of 300-3500 Hz, a bandwidth of just 3.2 kHz. This narrow bandwidth gives telephone calls their characteristic "muffled" sound, as compared to the rich wideband sound of high fidelity systems.
This is a problem for human speech, in which much of the information content, particularly in consonants, is in the frequencies above 3.2kHz and so is cut out by conventional telephone bandwidth. For example, in a telephone conversation, have you ever confused words like "see" and "fee"? The "f" and "s" sounds are easily confused (as are "t" and "d," "z" and "c," and many more) because their intelligibility is lost with when the higher frequencies are deleted by the telephone system.
G.722.1 provides 7kHz of audio bandwidth (50-7000 Hz), a vast improvement, and closer to FM radio quality than to traditional telephone quality; G.722.1C extends this to 14kHz, and G.719 covers the whole range of human hearing, to 20kHz.
Wideband audio is unanimously preferred when compared to narrowband audio quality; the whole audio experience when using wideband is far more natural and relaxing to the ears.
G.722.1 is also capable of excellent music reproduction at unprecedented low bit rates. It sounds far better than AM radio.

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Example G.722.1 applications:

  • Wideband IP telephony
  • Streaming audio (including music!) over the Internet
  • Video conferencing
  • Audio conferencing
  • Audio storage playback (recorders...)
  • Store-and-forward messaging (voice mail)
  • Audio-enabling your Web site

IP telephony, video conferencing, and audio conferencing all have very similar audio needs: high audio quality with low latency and complexity. In addition, the ability to change bit rate to accommodate channel requirements is necessary. G.722.1 allows the rate to change between the 24- and 32- kbps rates on any 20 ms frame boundary.
In streaming applications, low complexity and cost in the client is an absolute must. G.722.1 fulfils this requirement without sacrificing quality. Bit rates of 24- and 32 kbps enable clients to experience high-quality audio even on V.90 (56 kbps) modem connections.

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Hear G.722.1 for yourself.

Download these .WAV files to hear the quality of G.722.1 for yourself. If your browser has compatibility problems with audio downloads,, please try another browser.
(Note these sample files are uncompressed .WAV files playable on any computer. You don't need the G.722.1 codec to play them)

Speech files


3.5 kHz audio bandwidth, POTS toll quality

speech_3p5kHz_mulaw.wav (download size 114 KB)

7 kHz audio bandwidth, coded at 16 kbps using G.722.1 extension

speech_16kbps_siren.wav (download size 452 KB)

7 kHz audio bandwidth, coded at 24 kbps using G.722.1

speech_24kbps_g722p1.wav (download size 452 KB)

7 kHz audio bandwidth, coded at 32 kbps using G.722.1

speech_32kbps_g722p1.wav (download size 452 KB)

Music files


3.5 kHz audio bandwidth, POTS toll quality

music_3p5kHz_mulaw.wav (download size 72 KB)

7 kHz audio bandwidth, coded at 16 kbps using G.722.1 extension

music_16kbps_siren.wav (download size 286 KB)

7 kHz audio bandwidth, coded at 24 kbps using G.722.1

music_24kbps_g722p1.wav (download size 286 KB)

7 kHz audio bandwidth, coded at 32 kbps using G.722.1

music_32kbps_g722p1.wav (download size 286 KB)

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How do I get capabilities exchange for H.320, H.323 and H.324 systems?

In order for different vendors' equipment to interoperate using G.722.1, it is necessary to standardize the capability exchange and mode selection for G.722.1. These technical aspects for H.320, H.323, and H.324 systems have been defined by ITU-T Study Group 16.
Licensees will receive all information necessary to negotiate the use of G.722.1 per ITU-T standards.

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How do I use G.722.1 in SIP systems?

In order for different vendors' equipment to interoperate using G.722.1, it is necessary to standardize the capability exchange and mode selection for G.722.1. These technical aspects for SIP systems have been defined by RFC 5577.

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What audio quality does Polycom G.722.1C technology deliver?

Polycom G.722.1C passes the audio band between 50Hz and 14000 Hz, with excellent quality. Since Polycom G.722.1C is a transform-based codec, and not a speech-model based codec, it handles music and natural sounds just as well as speech, and does not "break up" on non-speech signals like many speech-model based codecs.
The 50 Hz to 14kHz bandwidth covers 100% of the energy in human speech - this is approximately the same bandwidth as offered by FM broadcast radio. This is also about the whole range that over-30 adults can hear (high-frequency hearing tends to decline with age).

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Can Polycom G.722.1C technology be used for stereo?

Yes, you can send two channels of Polycom G.722.1C audio to provide stereo. This requires double the bit rate of mono Polycom G.722.1C.
To preserve phase differences between the channels, we suggest that samples taken at the same instant should be stored or transported together, for example according to RFC 3551.

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Is Polycom's acoustic echo cancellation or noise reduction technology included with Polycom G.722.1C technology?

No. All of Polycom's audio codec licenses are for the audio codec technology only.

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How does Polycom G.722.1C technology compare to other "free" audio codecs?

Other "free" audio codecs that we are aware of either make use of technology on which the patents have expired (principally PCM and ADPCM codecs), in which case they require far higher bit rates (typically at least 128,000 bits/s) to approach Polycom G.722.1C technology's performance, or they use techniques about which there is dispute as to whether they infringe on patents held by third parties.
To our knowledge, Polycom's Siren technology is the only modern state-of-the-art audio coding technology offered on royalty-free terms.

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How does Polycom G.722.1C technology compare to other super-wideband audio codec standards, such as MPEG-4 AAC-LD, MPEG-4 AAC-LC, 3GPP eAAC+, and 3GPP AMR-WB+?

At a given bit rate Polycom G.722.1C technology provides audio quality and latency very similar to, and often better than, these other codecs, while consuming 1/4th to 1/20th of the CPU cycles they require. This makes it possible to operate Polycom G.722.1C with a lower-cost, lower-power processor, to operate more channels of Polycom G.722.1C on a given platform, or to free up processor cycles for other jobs such as video processing.
As part of the G.722.1 Annex C standardization process, the ITU extensively tested Polycom G.722.1C against MPEG-4 AAC LD. The ITU tests showed that the audio quality of Polycom G.722.1C is better than that of MPEG-4 AAC LD at all bit rates. Polycom G.722.1C also offers the same or lower latency compared to AAC LD, and much lower computational complexity.
For a given bit rate and processor cycle budget, Siren technology is by far the most efficient high-fidelity audio codec technology in the world today; this is why it was chosen as the international standard ITU-T Recommendation G.722.1 Annex C.
As well, Polycom G.722.1C is offered royalty-free; the others are not.
The document Polycom G.722.1C Information for Prospective Licensees has details on Polycom G.722.1C's audio quality and computation requirements, and summarizes the results of the independent tests of Polycom G.722.1C that were done in the ITU standardization process for G.722.1 Annex C.
You are invited to download a PC executable version of Polycom G.722.1C from Polycom's Web site. With this, you can test the quality of Polycom G.722.1C for yourself.

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Will Polycom help me integrate the Polycom G.722.1C software into my application?

No. We supply working ANSI C source code, but as we receive no royalty revenues we can't afford to help you with your implementation.

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Who can I contact at Polycom about Polycom codec technology?

Please send an email to audio.codec@polycom.com

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Where can I get a copy of the official G.722.1 Annex C standard?

The standard is formally known as ITU-T Recommendation G.722.1 Annex C, and is published by the International Telecommunications Union (ITU) in Geneva, Switzerland as a part of G.722.1.
You can obtain a copy from ITU at their Web site: http://www.itu.int/rec/T-REC-G.722.1-200505-I/en

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What Is the ITU?

The ITU is the International Telecommunications Union, the world's oldest international treaty organization, established in 1865 to set international standards for telecommunications in order to ensure worldwide interoperability. ITU members include almost every government in the world and many hundreds of commercial telecommunication operators and equipment manufacturers from around the globe. Today the ITU is part of the United Nations. The ITU Web site is http://www.itu.int.

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Where can I get the executable file of Polycom G.722.1C technology for evaluation?

How can I verify compatibility between my implementation and G.722.1 or G.722.1 Annex C?

ITU-T Recommendation G.722.1 Annex A specifies the packet format, capability identifiers, and capability parameters for G.722.1 and G.722.1C and they can use that document to verify their design. For real-time testing, they can call Polycom's systems with G.722.1C to know if there are the interoperability problems.

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Is the reference G.722.1/G.722.1 code in Little Endian or Big Endian Mode or both?

It is "Little Endian", i.e.: the low-order byte of the number is stored in memory at the lowest address, and the high-order byte at the highest address.

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What is the delay of the G.722.1/G.722.1C codec?

The frame length is 20 ms. The total stated delay of 40 ms is the sum of two parts:

  • 20 ms due to frame buffering
  • 20 ms one frame look-ahead delay – the transform window

To make this clear, suppose your audio is already packetized into 20 ms. Then G.722.1/G.722.1C will only add an additional frame delay – 20 ms ( NOT 40 ms )

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Why does G.722.1/G.722.1C have a longer latency than G.722?

G.722.1 is what's known as a "transform codec." Any transform codec (Fourier transform or comparable) needs enough of a temporal sample to be able to do a frequency analysis. In G.722.1/G.722.1C, 20ms was chosen as the best compromise for this application.
G.722 is fundamentally different – it's an ADPCM codec, and operates sample by sample so the latency is lower. Latency is actually boosted a bit by some pre-filtering within G.722 as it's really two ADPCM codecs, and operates sample by sample on two separate bands – the low half and the high half.
The tradeoff is that G.722 requires two to three times the data rate. So as is always the case in codec selection, choosing a codec will be a task of prioritizing requirements. For VoIP, we haven't seen the incremental delay of G.722.1 presenting a problem, especially considering its higher data efficiency, and it's in the middle of the field with other codecs; AMR and AMR-WB, SILK, ISAC, and others present similar tradeoffs between latency and efficiency. For more information on this, please refer to the white paper "VoIP to 20 kHz: Codec Choices for High Definition Voice Telephony," which can be found athttp://www.polycom.com/global/documents/whitepapers/codecs_white_paper.pdf  

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How do these delays compare to MP3?

MP3 (MPEG-1/2 Layer-3) has an algorithmic delay of 2591 samples. For the sample rate of 48kHz, it corresponds to about 54ms. In case of 32kHz, it's about 81ms.

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What is the total data rate when G.722.1/G.722.1C is encapsulated within RTP?

The bit rate will vary once encapsulated in RTP; the Siren payload is not treated differently than another codec or stream, and so consequently the tabulation will be specific to the network and the application. However, the RTP payload for G.722.1 and G.722.1C is specified in ITU-T Recommendation G.722.1 Annex A, and also specified in IETF RFC 5577.

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What are the correct signal formats for running the G.722.1/G.722.1C simulator on a PC?

The input signal to G.722.1C should be 16-bit PCM format with no header (usually named as *.pcm). The *.wav format has a header and should be converted into 16-bit PCM format for encoding. The output of the codec is the same 16-bit PCM signal; this can not be played with Windows Media Player.
The input signal to G.722.1C should be 16-bit PCM format with no header (usually named as *.pcm). The *.wav format has a header and should be converted into 16-bit PCM format for encoding. The output of the codec is the same 16-bit PCM signal; this can not be played with Windows Media Player.
To correctly code the file, perform the following steps:

  • Convert the *.wav file into the *.pcm format. This can be done using commercial software such as CoolEdit (now Adobe Audition).
  • Encode and decode the file.
  • Convert the decoder output into the *.wav format to play it with Windows Media Player,

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What is the difference between Siren7 and G.722.1?

G.722.1 is the formal name for the codec, as approved by the ITU. Siren7 is an informal name which refers to the same general codec.

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Does my codec have to produce a bit-exact match?

Officially, ITU-T requires a bit-exact match. This is the reason for supplying a "C" implementation to the public. Polycom does not monitor implementations for this, but if they are not bit-exact there is a high probability that they will function poorly or not at all.

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When does Polycom use the RTP header for G.722.1/G.722.1C?

For H.323 and SIP Polycom uses the RTP format. For H.320 it is in the exact same format, but without a RTP header

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Since it is royalty-free do I still need to take a license?

Yes. Parties who want to use G.722.1/G.722.1C/G.719 must take a license and comply with the terms of the license. If you use G.722.1/G.722.1C/G.719 without a license you will be infringing on our patents and potentially our copyrights.

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Is G.722.1/G.722.1C/G.719 software really completely free?

Yes. You don't have to pay Polycom one penny to use G.722.1/G.722.1C/G.719. However, you do have to obtain a license from Polycom and comply with its terms.

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What are the terms on the free license?

The main requirements are:

  • Your implementation must conform to and be interoperable with the relevant of ITU T Recommendation G.722.1, G.722.1 Annex C, and G.719 standard. Our goal is to encourage interoperability so that our customers will enjoy the benefits of super-wideband audio quality)
  • Don't sue us over your own patents.

Please read the license itself carefully for full details

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Does this amount to a blanket cross-license of all my patents in exchange for the Siren/Siren14/G.719 license?

No. Polycom respects the patent rights of others. The termination provision is to provide a more appropriate setting in potential cross-licensing discussions.

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Do I need a license from someone else besides Polycom?

As to ITU T Recommendation G.722.1, and G.722.1 Annex C to our best knowledge and belief, no. Our belief is that all patents on Siren and Siren 14 technology are owned by Polycom. As of this writing (March 2015) no other party has claimed otherwise, even informally. As to G.719, Ericsson Corporation has certain rights in G.719, so a license from Ericsson would also be required for G.719.
However, as with any technology, it is always possible that third parties may claim, correctly or not, to have rights that are infringed by Siren/Siren14/G.719 technology. Ultimately only a court of law can decide the validity of such claims. Polycom offers no guarantees that no one will make such a claim in the future, and will not compensate licensees for losses resulting from any such claims. Licensees are responsible for their own risks in this regard.

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Are other G.722.1 codecs also free?

Both G.722.1 and G.722.1 Annex C technologies are now offered on royalty-free terms as described above. The license agreement with Polycom covers Polycom's IP as implemented within both of these standards, as well as within ITU-T G.719, the new fullband (20kHz) codec.

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Why is Polycom offering G.722.1/G.722.1C/G.719 technology royalty-free?

Polycom, as the world's leading provider of unified collaborative communications solutions including video conferencing and high-end teleconferencing equipment, seeks to expand our market by making telecommunications more effective, natural, and valuable.
By offering G.722.1/G.722.1C/G.719 on royalty-free terms, we hope to encourage the widespread implementation and use of super-wideband quality audio that is standards-based and interoperable across all vendors, so that our customers will enjoy the ultimate in clear, natural-sounding audio.
Polycom is proud of our long heritage of leadership in media coding technology and standardization, and by making G.722.1/G.722.1C/G.719 available royalty-free we have an opportunity to demonstrate our superior technology and industry leadership to the widest possible audience.

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Where can I get more technical information?

Technical Contact

For additional technical information please email: audio.codec@polycom.com