The numbering of the wideband ITU audio codecs is sometimes confusing. There are three principal codecs, which are unrelated, but all carrying the G.722 label. G.722 is the original 7 kHz codec, using ADPCM and operating at 48 – 64 kbit/s. G.722.1, another 7 kHz codec, operates at half the data rate while delivering comparable or better quality than G.722, but is a transform-based codec. G.722.1 Annex C is very similar to G.722.1, but provides twice the audio bandwidth, 14 kHz. And G.722.2, which operates onwideband speech and delivers very low bitrates, is an ACELP-based algorithm.
List
G.722
G.722.1
G.722.1 Annex C
G.722.2
For more info on CUCM Codec, see "Region" topics under CUCM System Guide, http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmsys/CUCM_BK_CD2F83FA_00_cucm-system-guide-90/CUCM_BK_CD2F83FA_00_system-guide_chapter_0101.html#CUCM_RF_RE6237E1_00
Regions
Regions provide capacity controls for Cisco Unified Communications Manager multi-site deployments where you may need to limit the bandwidth for individual calls that are sent across a WAN link, but where you want to use a higher bandwidth for internal calls. Additionally, the system uses regions also for applications that only support a specific codec; for example, an application that only uses G.711. Use regions to specify the maximum transport-independent bit rate that is used for audio and video calls within a region and between regions; in this case, codecs with higher bit rates do not get used for the call.
Cisco Unified Communications Manager prefers codecs with better audio quality. For example, despite having a maximum bit rate of 32 kb/s, G.722.1 sounds better than some codecs with higher bit rates, such as G.711, which has a bit rate of 64 kb/s.
When you configure the maximum audio bit rate setting in the Region Configuration window (or use the service parameter in the Service Parameter Configuration window), this setting serves as a filter. When an audio codec is selected for a call, Cisco Unified Communications Manager takes the matching codecs from both sides of a call leg, filters out the codecs that exceed the configured maximum audio bit rate, and then picks the preferred codec among the codecs that are remaining in the list.
The audio codec preference feature orders the audio preference in the following table for the default low-loss case by sound quality, and the table adds a separate preference list for the lossy case.
For calls made between Cisco Unified Communications Manager and previous versions of Cisco Unified Communications Manager over SIP intercluster trunks, the Cisco Unified Communications Manager that makes the SDP Answer chooses the codec. Because of SIP Delayed Offer support, the Cisco Unified Communications Manager that initiates or resumes the call is the one that makes the SDP Answer, and hence, it is the one that determines the codec for the call.
For audio calls that involve H.323 intercluster trunks, Cisco Unified Communications Manager uses the preference list of codecs in the previous table only if both sides of the call run Cisco Unified Communications Manager 8.6(1). If both sides of the call do not run Cisco Unified Communications Manager 8.6(1), the codec list from the following table gets used.
For audio and video calls, Cisco Unified Communications Manager uses the preference order of codecs in the following table.
Supported Audio Codecs
Cisco Unified Communications Manager supports video stream encryption and various audio/video codecs, such as G.722. Cisco Unified Communications Manager supports the following audio codecs:
- G.711-The most commonly supported codec, used over the public switched telephone network.
- G.722-G.722 is wideband codec that is always preferred by Cisco Unified Communications Manager over G.711, unless G.722 is disabled. Audio codec often used in video conferences. See the Codec Usage of the Cisco Unified IP Phones chapter for a detailed discussion of the Advertise G.722 Codec enterprise parameter, which determines whether Cisco Unified IP Phones will advertise the G.722 codec to Cisco Unified Communications Manager.
- G.722.1-G.722.1 is a low-complexity wideband codec operating at 24 and 32 kb/s. The audio quality approaches that of G.722 while using at most half the bit rate. As it is optimized for both speech and music, G.722.1 has slightly lower speech quality than the speech-optimized iSAC codec. G.722.1 is supported for SIP and H.323 devices.
- G.723.1-Low-bit-rate codec with 6.3 or 5.3 kb/s compression for Cisco IP Phone 12 SP+ and Cisco IP Phone 30 VIP devices.
- G.728-Low-bit-rate codec that video endpoints support.
- G.729-Low-bit-rate codec with 8-kb/s compression that is supported by Cisco Unified IP Phone 7900. Typically, you would use low-bit-rate codecs for calls across a WAN link because they use less bandwidth. For example, the factory default intraregion maximum audio bit rate is 64 kbps, while the factory default interregion maximum audio bit rate is 8 kbps.
- GSM--The global system for mobile communications (GSM) codec. GSM enables the MNET system for GSM wireless handsets to operate withCisco Unified Communications Manager. Assign GSM devices to a device pool that specifies 13 kb/s as the audio codec for calls within the GSM region and between other regions. Depending on device capabilities, this includes GSM EFR (enhanced full rate) and GSM FR (full rate).
- L16-Uncompressed 16-bit linear pulse-code modulation (PCM) encoded audio with a 16-kHz sampling rate provides wideband audio at 256 kb/s. Works with phones with handsets, acoustics, speakers, and microphones that can support high-quality audio bandwidth, such as the Cisco Unified IP Phone7900 Series.
- AAC-LD (mpeg4-generic)-Advanced Audio Coding-Low Delay (AAC-LD) is a super-wideband audio codec that provides superior sound quality for voice and music. This codec provides equal or improved sound quality over older codecs, even at lower bit rates.AAC-LD (mpeg4-generic) is supported for SIP devices, in particular, Cisco TelePresence systems.
- AAC-LD (MP4A-LATM)-Advanced Audio Coding-Low Delay (AAC-LD) Low-overhead MPEG-4 Audio Transport Multiplex (LATM) is a super-wideband audio codec that provides superior sound quality for voice and music. This codec provides equal or improved sound quality over older codecs, even at lower bit rates.AAC-LD (MP4A-LATM) is supported for SIP devices including Tandberg and some third-party endpoints.
Note
AAC-LD (mpeg4-generic) and AAC-LD (MPA4-LATM) are not compatible.
- iSAC-Internet Speech Audio Codec (iSAC) is an adaptive wideband audio codec, specially designed to deliver wideband sound quality with low delay in both low and medium-bit rate applications.Using an adaptive bit rate of between 10 and 32 kb/s, iSAC provides audio quality approaching that of G.722 while using less than half the bandwidth. In deployments with significant packet loss, delay, or jitter, such as over a WAN, iSAC audio quality is superior to that of G.722 due to its robustness.iSAC is supported for SIP and SCCP devices. The Cisco Unified Communications Manager IP Voice Media Streaming App (IPVMSApp), which includes Media Termination Point, Conference Bridge, Music on Hold Server, and Annunciator does not support iSAC. MGCP devices are not supported.
- iLBC-Internet Low Bit Rate Codec (iLBC) provides audio quality between that of G.711 and G.729 at bit rates of 15.2 and 13.3 kb/s, while allowing for graceful speech quality degradation in a lossy network due to the speech frames being encoded independently. By comparison, G.729 does not handle packet loss, delay, and jitter well, due to the dependence between speech frames.iLBC is supported for SIP, SCCP, H323, and MGCP devices.
Note |
H.323 Outbound FastStart does not support the iLBC codec.
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- AMR-Adaptive Multi-Rate (AMR) codec is the required standard codec for 2.5G/3G wireless networks based on GSM (WDMA, EDGE, GPRS). This codec encodes narrowband (200-3400 Hz) signals at variable bit rates ranging from 4.75 to 12.2 kb/s with toll quality speech starting at 7.4 kbps.AMR is supported only for SIP devices.
- AMR-WB-Adaptive Multi-Rate Wideband (AMR-WB) is codified as G.722.2, an ITU-T standard speech codec, formally known as Wideband coding of speech for about 16 kb/s. This codec is preferred since it provides excellent speech quality due to a wider speech bandwidth of 50 Hz to 7000 Hz compared to other narrowband speech codecs.AMR-WB is supported only for SIP devices.
Note |
AMR-WB is preferred by Cisco Unified Communications Manager over AMR and other supported codecs, G.711 in particular.
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The total bandwidth that is used per call stream depends on the audio codec type as well as factors such
Note |
Each call includes two streams, one in each direction.
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Note |
For information on bandwidth usage for each codec, see the Cisco Unified Communications Solution Reference Network Design (SRND) for the current release of Cisco Unified Communications Manager.
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Audio Codec
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Bandwidth Used for Data Packets Only (Fixed Regardless of Packet Size)
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Bandwidth Used Per Call (Including IP Headers) With 30-ms Data Packets
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Bandwidth Used Per Call (Including IP Headers) With 20-ms Data Packets
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|---|---|---|---|---|---|
G.711
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64 kb/s
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80 kb/s
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88 kb/s
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G.722
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64 kb/s
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80 kb/s
|
88 kb/s
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G.722.1
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24 kb/s
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Not applicable
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40 kb/s
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G.722.1
|
32 kb/s
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Not applicable
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48 kb/s
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iSAC
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32 kb/s
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32 kb/s
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G.723.1
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6.3 or 5.3 kb/s
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24 kb/s
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Not applicable
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G.728
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16 kb/s
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26.66 kb/s for G.728
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iLBC
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15.2 or 13.3 kb/s
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24 kb/s for iLBC
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G.729
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8 kb/s
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24 kb/s
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32 kb/s
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L16
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256 kb/s
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272 kb/s
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280 kb/s
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AAC-LD (mpeg4-generic)
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256 kb/s
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272 kb/s
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AAC-LD (MP4A-LATM)
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128 kb/s
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Not applicable
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156 kb/s1.
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AAC-LD (MP4A-LATM)
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64 kb/s
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Not applicable
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88 kb/s
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AAC-LD (MP4A-LATM)
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56 kb/s
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Not applicable
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80 kb/s
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AAC-LD (LATM)
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48 kb/s
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Not applicable
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72 kb/s
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AAC-LD (MP4A-LATM)
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32 kb/s
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Not applicable
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56 kb/s
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AAC-LD (MP4A-LATM)
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24 kb/s
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Not applicable
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48 kb/s
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GSM (Global system for mobile communications)
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13 kb/s
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29 kb/s
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37 kb/s
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1 AAC-LD (MP4A-LATM) does not specify the packetization period (20 ms or 30 ms) in SDP (it assumes the maximum overhead of 24K, which is in 20 ms)
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