Codec Supports on CUCM 10.x

CUCM System Guides provides detail list of the codecs as well as some of their characteristics. (See CUCM System Guides 10.x here http://goo.gl/VhVCsp)

Codec Preference Order within CUCM and over SIP Intercluster Trunk

One of the key takeaway is the codec preference on Lossless vs Lossy links. (Refer to table 2 on the page)

Link Loss Type = Lossless

G.722 64k-64 kb/s
iSAC-32 kb/s
AAC-LD (MP4A-LATM)-32 kb/s

Link Loss Type = Lossy

iSAC-32 kb/s
AAC-LD (MP4A-LATM)-32 kb/s
G.722 64k-64 kb/s

"....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...."

Table 2 Audio Codec Preference Order for Cisco Unified Communications Manager

If Low Loss Is Configured for Link Loss Type	If Lossy Is Configured for Link Loss Type
AMR-WB-24 kb/s	AMR-WB-24 kb/s
AMR-13 kb/s	AMR-13 kb/s
AAC-LD (MP4A-LATM)-128 kb/s	AAC-LD (MP4A-LATM)-128 kb/s
AAC-LD (mpeg4-generic)-64 kb/s	AAC-LD (mpeg4-generic)-64 kb/s
AAC-LD (MP4A-LATM)-64 kb/s	AAC-LD (MP4A-LATM)-64 kb/s
AAC-LD (MP4A-LATM)-56 kb/s	AAC-LD (MP4A-LATM)-56 kb/s
L16-256 kb/s	L16-256 kb/s
AAC-LD (MP4A-LATM)-48 kb/s	AAC-LD (MP4A-LATM)-48 kb/s
G.722 64k-64 kb/s	iSAC-32 kb/s
iSAC-32 kb/s	AAC-LD (MP4A-LATM)-32 kb/s
AAC-LD (MP4A-LATM)-32 kb/s	G.722 64k-64 kb/s
G.722.1 32k-32 kb/s	G.722.1 32k-32 kb/s
G.722 -56 kb/s	G.722 -56 kb/s
G.722.1-24 kb/s	G.722.1-24 kb/s
G.722-48 kb/s	G.722-48 kb/s
AAC-LD (MP4A-LATM)-24 kb/s	AAC-LD (MP4A-LATM)-24 kb/s
G.711 mu-law 64 k-64 kb/s	G.711 mu-law 64 k-64 kb/s
G.711 A-law 64k-64 kb/s	G.711 A-law 64k-64 kb/s
G.711 mu-law 56k-56 kb/s	G.711 mu-law 56k-56 kb/s
G.711 A-law 56k-56kb/s	G.711 A-law 56k-56kb/s
iLBC-16 kb/s	iLBC-16 kb/s
G.728-16 kb/s	G.728-16 kb/s
GSM Enhanced Full Rate-13 kb/s	GSM Enhanced Full Rate-13 kb/s
GSM Full Rate-13 kb/s	GSM Full Rate-13 kb/s
G.729b-8 kb/s	G.729b-8 kb/s
G.729ab-8 kb/s	G.729ab-8 kb/s
G.729-8 kb/s	G.729-8 kb/s
G.729a-8 kb/s	G.729a-8 kb/s
GSM Half Rate-7 kb/s	GSM Half Rate-7 kb/s
G.723.1-7 kb/s	G.723.1-7 kb/s

Codec Preference Order on H323 Intercluster Trunk

Table 3 Audio Codec Preference Order for H.323 Intercluster Trunks If Both Sides of Call Do Not Support Cisco Unified Communications Manager 8.5(1))

If Low Lossy Is Configured for Link Loss Type	If Lossy Is Configured for Link Loss Type
---	iLBC-16 kb/s
AAC-LD (mpeg4-generic)-256 kb/s	AAC-LD (mpeg4-generic)-256 kb/s
L16-256 kb/s	L16-256 kb/s
G.722.1 24k-24 kb/s	G.722.1 24k-24 kb/s
G.722.1 32k-32 kb/s	G.722.1 32k-32 kb/s
G.722 64k-64 kb/s	G.722 64k-64 kb/s
G.711 mu-law 64k-64 kb/s	G.711 mu-law 64k-64 kb/s
G.711 A-law 64k-64 kb/s	G.711 A-law 64k-64 kb/s
G.722 56k-56 kb/s	G.722 56k-56 kb/s
G.711 mu-law 56k-56 kb/s	G.711 mu-law 56k-56 kb/s
G.711 A-law 56k-56 kb/s	G.711 A-law 56k-56 kb/s
G.722 48k-48 kb/s	G.722 48k-48 kb/s
iLBC-16 kb/s	---
G.728-16 kb/s	G.728-16 kb/s
GSM Enhanced Full Rate-13 kb/s	GSM Enhanced Full Rate-13 kb/s
GSM Full Rate-13 kb/s	GSM Full Rate-13 kb/s
G.729b-8 kb/s	G.729b-8 kb/s
G.729ab-8kb/s	G.729ab-8kb/s
G.729-8 kb/s	G.729-8 kb/s
G.729a-8 kb/s	G.729a-8 kb/s
GSM Half Rate-7 kb/s	GSM Half Rate-7 kb/s
G.723.1-7 kb/s	G.723.1-7 kb/s

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.

1. 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.
2. 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.
3. G.728-Low-bit-rate codec that video endpoints support.
4. 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.
5. 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).
6. 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.
7. 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.
8. 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.
9. 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.
10. 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.
11. 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.
12. 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.

Bandwidth Required per Call

Table 4 Bandwidth Used Per Call by Each Codec Type in IPv4

Audio Codec	Bandwidth Used for Data Packets Only (Fixed Regardless of Packet Size)	Bandwidth Used Per Call (Including IP Headers) With 30-ms Data Packets	Bandwidth Used Per Call (Including IP Headers) With 20-ms Data Packets
G.711	64 kb/s	80 kb/s	88 kb/s
G.722	64 kb/s	80 kb/s	88 kb/s
G.722.1	24 kb/s	Not applicable	40 kb/s
G.722.1	32 kb/s	Not applicable	48 kb/s
iSAC	32 kb/s	32 kb/s
G.723.1	6.3 or 5.3 kb/s	24 kb/s	Not applicable
G.728	16 kb/s	26.66 kb/s for G.728
iLBC	15.2 or 13.3 kb/s	24 kb/s for iLBC
G.729	8 kb/s	24 kb/s	32 kb/s
L16	256 kb/s	272 kb/s	280 kb/s
AAC-LD (mpeg4-generic)	256 kb/s	272 kb/s
AAC-LD (MP4A-LATM)	128 kb/s	Not applicable	156 kb/s1.
AAC-LD (MP4A-LATM)	64 kb/s	Not applicable	88 kb/s Note    See footnote 1.
AAC-LD (MP4A-LATM)	56 kb/s	Not applicable	80 kb/s Note    See footnote 1.
AAC-LD (LATM)	48 kb/s	Not applicable	72 kb/s Note    See footnote 1.
AAC-LD (MP4A-LATM)	32 kb/s	Not applicable	56 kb/s Note    See footnote 1.
AAC-LD (MP4A-LATM)	24 kb/s	Not applicable	48 kb/s Note    See footnote 1.
GSM (Global system for mobile communications)	13 kb/s	29 kb/s	37 kb/s

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)