In this section:
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.
The following features are not supported on the
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platform:- G.722.1
- EFR
- T.38
Audio Codecs Supported (Transcoding and Pass-Through)
Audio Codecs Supported (Transcoding and Pass-Through)
(PSX) Codec Selections | Coding Rates | Packet Sizes (bytes) |
|---|---|---|
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 |
AMR-Octet Aligned | Same for all AMR | 20, 40, 60 |
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 | 20, 40, 60, 80, 100 |
G.729A+B | 8 | 20, 40, 60, 80, 100 |
iLBC | 13.3, 15.2 | 30, 60 |
iLBC with Silence Suppression | 13.3, 15.2 | 30, 60 |
| Opus |
| 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 | Packet Sizes (Bytes) |
|---|---|---|
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-2/8000 | 82 | 20 |
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) | 13.2 | 20, 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-RTA | 8, 16 | 20, 40, 60 |
| SILK | 8, 12, 16, 24 | 20, 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 Unable to show "metadata-from": No such page "_space_variables"
also renegotiates the media during a call. The Unable to show "metadata-from": No such page "_space_variables"
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
Opus transcoding is not supported on
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platforms.The
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platforms support the Opus audio codec in accordance with RFC 6716 and draft-ietf-payload-rtp-opus-01 (see 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)
Currently on
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, Opus codec does not support DTX on the transmitter side (i.e. the encoder does not go into DTX mode during silence or background noise) nor FEC in neither transmitter or receiver side (i.e. the encoder does not encode FEC frames and the decoder does not attempt to recover any lost packet from FEC frames in future packets).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.
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 Unable to show "metadata-from": No such page "_space_variables"
:- 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.
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
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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 are enhanced to support this feature:| Parameter | Current Behavior | Enhanced Behavior |
|---|---|---|
Maxaveragebitrate | min (route PSP, 20kbit/s) | min (offer/answer of peer, route PSP, 20kbit/s) |
Useinbandfec | Route-Config | min (offer/answer of peer, route PSP) |
usedtx | Route-Config | min (offer/answer of peer, route PSP) |
usecbr | Route-Config | Constant bit rate if either peer requests cbr=1 OR route is configured for cbr=1 |
Overview
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