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Audio Codec Support

This section summarize audio codec support with and without (pass-through) transcoding selectable on the

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.

Audio codec relay is supported in SIP-SIP, SIP-H.323 and H.323-H.323 interworking calls.

Note

The 

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does not support transcoding the following codecs:

  • EFR
  • G.722.1
  • T.38

Audio Codecs Supported (Transcoding and Pass-Through)

Audio Codecs Supported (Transcoding and Pass-Through)

(PSX) Codec Selections

Coding Rates
(Kbps)

Packetization Periods
(ms)

AMR-Bandwidth Efficient

Variable - mix of 4.75, 5.15, 5.90, 6.70, 7.40, 7.95, 10.20, 12.20

20, 40, 60, 80

AMR-Octet Aligned

Same for all AMR

20, 40, 60, 80

EFR

12.2

20, 40, 60
EVRC

Variable (171, 80, or 16 bit samples per 20 ms)

20, 40, 60

EVRC0

Variable (171, 80, or 16 bit samples per 20 ms)

20

EVRCB0

4.8, 5.8, 6.2, 6.6, 7.0, 7.5, 8.5, 9.3

20
EVRCB

Variable (171, 80, 40, or 16 bit samples per 20 ms)

20, 40, 60

G.711

64

10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60

G.711 with Silence Suppression

64

10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60

G.722

48, 56, 64

10, 20, 30, 40

G.722.1

16, 24, 32

20, 40, 60, 80

G.722.2 (AMRWB-Bandwidth Efficient)

6.6, 8.85, 12.65, 14.25, 15.85, 18.25, 19.85, 23.05, or 23.85

20, 40, 60, 80, 100

G.722.2 (AMRWB-Octet Aligned)

Same for all AMR-WB

20, 40, 60, 80, 100

G.723.1

5.3, 6.3

30, 60, 90, 120, 150

G.723.1A

5.3, 6.3

30, 60, 90, 120, 150

G.726

32

10, 20, 30, 40

G.726 with Silence Suppression

32

10, 20, 30, 40

G.729A (compatible with G.729)

8

10, 20, 30, 40, 50, 60

G.729A+B

8

10, 20, 30, 40, 50, 60
iLBC, iLBC-SS15.220, 40, 60
13.330, 60
Opus
  • 6 to 20 (transcoded)
  • 6 to 510 (pass-through)

10, 20, 30, 40, 50, 60

T.38 (version 0)

up to 14.4

 
T.38 (version 3)up to 33.6 

Audio Codecs Supported (Pass-Through Only)

Audio Codecs Supported (Pass-Through Only)

(PSX) Codec Selections

Coding Rates
(Kbps)

Packetization Periods
(ms)

AMR-CRC

Same for all AMR

20, 40, 60

AMR-CRC Robust Sorting

Same for all AMR

20, 40, 60

AMR-CRC-Interleaving-Robust Sorting

Same for all AMR

20, 40, 60

AMR-Interleaving

Same for all AMR

20, 40, 60

AMR-Interleaving-Robust Sorting

Same for all AMR

20, 40, 60

AMR-Robust Sorting

Same for all AMR

20, 40, 60

Broadvoice

16, 32

5, 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60

Broadvoice with FEC

32

15, 20, 25, 30, 35, 40, 45, 50, 55, 60
Dolby DVC-2/80008220

EVRC1

Variable (171, 80, or 16 bit samples per 20 ms)

20, 40, 60

EVRC1 FR

8.55

20, 40, 60

EVRCB1

Variable (171, 80, 40, or 16 bit samples per 20 ms)

20, 40, 60

EVRCB1 FR

8.55

20, 40, 60

G722 with Silence Suppression

48, 56, 64

10, 20, 30, 40

G.722.2 (AMRWB-CRC)

Same for all AMR-WB

20, 40, 60, 80, 100

G.722.2 (AMRWB-CRC-Interleaving)

Same for all AMR-WB

20, 40, 60, 80, 100

G.722.2 (AMRWB-CRC-Robust Sorting)

Same for all AMR-WB

20, 40, 60, 80, 100

G.722.2 (AMRWB- CRC-Interleaving-Robust_Sorting)

Same for all AMR-WB

20, 40, 60, 80, 100

G.722.2 (AMRWB-Interleaving)

Same for all AMR-WB

20, 40, 60, 80, 100

G.722.2 (AMRWB-Interleaving-Robust Sorting)

Same for all AMR-WB

20, 40, 60, 80, 100

G.722.2 (AMRWB-Robust Sorting)

Same for all AMR-WB

20, 40, 60, 80, 100

G.728

16

10, 20, 30, 40, 50, 60, 70, 80,

90, 100, 110, 120, 130, 140, 150

G.728 with Silence Suppression

16

10, 20, 30, 40, 50, 60, 70, 80,

90, 100, 110, 120, 130, 140, 150

GSM (full rate)13.220, 40, 60

iSAC HD codec (pass-through and direct media)

10 to 32

30, 60

L16

128

5, 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60
MS-RTA8, 1620, 40, 60
SILK8, 12, 16, 2420, 40, 60, 80, 100

Speex

8, 16, 32

20, 40, 60

Speex with FEC

8, 16

20, 40, 60

The

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selects codecs on a call-by-call basis, and negotiates codec use with destination gateway during initial call setup. The 
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also renegotiates the media during a call. The 
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defines configuration parameters such as audio codec, packet size, and TOS to apply to individual call legs.

For direct media connections, bandwidth and policing requirements do not apply. Audio streams must have different IP port numbers, but may have the same or different remote IP addresses. Audio streams for each call leg can be allocated on the same or different IP interface.

Opus Codec Support

Note

Opus transcoding is not supported on

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and
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platforms.

The 

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,
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, 
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and
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platforms support the Opus audio codec in accordance with RFC 6716 and draft-ietf-payload-rtp-opus-01 (refer to Supported Standards page). There are no licensing requirements for this codec.

Opus is an open, royalty-free, highly versatile audio codec consisting of a combination of SILK (LPC) and CELT (Constrained-Energy Lapped Transform) codecs. At any given point of time the LP layer, the MDCT (CELT) layer or both may be active. Opus is used for the following applications:

  • VoIP and video conferencing
  • Music/video streaming and storage
  • Remote music jamming
  • Wireless speakers/headphones/microphone
  • Audio books
  • Virtualization/sound servers

Supported Opus Features

The 

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platforms include the following Opus functionality:

  • Input sampling rates of 8k Hz (NB) and 16k Hz (WB) are supported on the IDP interface (encoder input and decoder output)
  • Opus-to-Opus pass-through calls
    • Encoding output bandwidths of 8, 12, 16, 24 and 48 kHz
    • 6 kbps to 510 kbps bit rates
  • Opus transcoded calls
    • Max bandwidth of 16 kHz
    • Max bit rate of 20 kbps
  • Mono mode for both pass-through transcoded calls
  • Stereo mode for Opus-to-Opus pass-through.
  • Inband FEC mode
  • Variable bit rate (VBR)
  • DTX mode
  • Single channel mode is supported
  • Fax/Modem tone (FMTD) detection is not supported in Opus leg
  • DTMF (Inband DTMF detection is not guaranteed)

On 

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, Opus DTX is fully supported on both transmitter and receiver sides; however, packets are generated approximately every 200 milliseconds during DTX period.

 

Note

The following Opus attributes cannot be controlled by the operator:

  • Maxplaybackrate
  • Sprop-maxcapturerate
  • Stereo
  • sprop-stereo

UXPAD Operational Modes

As with other currently-supported compression codecs, the

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supports Opus in UXPAD only. The following UXPAD operational modes are supported by the
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:

  • WB to WB transcoding using two UXPADs connected back to back over IDP-WB format. The wideband voice features should be preserved while performing the transcoding in two stages using PADs, the intermediate IDP should always be wideband.
  • WB to NB transcoding (except for G.711-RTP) using two UXPAD connected back to back over IDP-NB format.
  • WB to G.711-RTP transcoding using single UXPAD.
  • Opus to TNAPAD to play out any tones or announcements.
  • Opus-to-Opus transcoding using two UXPADs connected back-to-back over IDP-WB format. This is used only for Opus-to-Opus transcoding that may occur due to existing Packet Service Profile (PSP) flags:
    • Transcode always
    • Conditional Transcoding > Conditions in addition to “No common Codec”:
      • Different ptime/Silence Suppression,
      • Different DTMF relay.
Note

Any up-sampling or down-sampling may introduce some noise and degrade speech quality. In addition, wideband speech quality is lost if down-sampling is done for wideband-to-narrowband transcoding. However, up-sampling / down-sampling is required for interworking

UXPAD Support of Opus

DTX and Inband FEC Support

The 

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supports DTX on the transmitter side and Inband FEC on the transmitter and the receiver side for Opus codec. The following parameters support this feature:

ParameterBehavior
Maxaveragebitratemin (offer/answer of peer, route PSP, 20kbit/s)
Useinbandfecmin (offer/answer of peer, route PSP)
usedtxmin (offer/answer of peer, route PSP)
usecbrConstant bit rate if either peer requests cbr=1 OR route is configured for cbr=1

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