Audio Codecs

 

Audio Codec Support

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

.

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

Audio Codecs Supported (Transcoding and Pass-Through)

The 

 supports the following narrowband (NB) and wideband (WB) audio codecs for transcoding and pass-through: 

Audio Codecs Supported (Pass-Through Only)

The 

 supports the following narrowband (NB) and wideband (WB) audio codecs for pass-through:

The

selects codecs on a call-by-call basis, and negotiates codec use with destination gateway during initial call setup. The  also renegotiates the media during a call. The  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.

EVS Pass-through Support

The SBC supports use of the Enhanced Voice Services (EVS) codec in pass-through mode. Pass-through of the EVS codec from one call leg to another can occur when EVS is configured in both the ingress and egress PSPs, based on the outcome of SDP offer-answer procedures.

In pass-through mode, all EVS codec parameters in Primary mode as well as AMRWB-IO mode are relayed to the egress peer. 

The SBC drops any unknown parameters received in the a=fmtp: line in the SDP offer or answer, as well as any known EVS codec parameters that have values other than the range defined in the TS 26.114 standard. 

To enable support of EVS, you can configure codec entries that specify EVS as the codec type. The codec entry configuration allows the SBC to restrict certain parameters, like bit-rate, according to the operator's requirements. The configurable options for an EVS-based codec entry include:

• Packet size 20-100ms supported - default 20 ms
• Dynamic Preferred RTP Payload Type 0-127 supported - default 96
• Min Bit Rate - Supported rates are: 5.9, 7.2,8, 9.6, 13.2, 16.4, 24.4, 32, 48, 64, 96, and 128, but must be less than or equal to Max Bit Rate - default 5.9
• Max Bit Rate - Supported rates are: 5.9, 7.2,8, 9.6, 13.2, 16.4, 24.4, 32, 48, 64, 96, and 128, but must be greater than or equal to Min Bit Rate - default 128
  The configured minimum and maximum for bit rate defines the range the SBC uses in its offer/answers during SDP negotiation.
  
• DTMF Relay - Supported values are: None (default), Out Of Band, Rfc2833, Either Oob Or Rfc2833, Both Oob And2833.
• DTMF Remove Digits - Supported values are Disable and Enable (default).

The configured Codec Entries are then incorporated in Packet Service Profiles (PSP) and assigned to enable EVS support.

Opus Codec Support

The 

, , , , and 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 

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 

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

UXPAD Operational Modes

As with other currently-supported compression codecs, the

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

• WB to WB transcoding using two UXPADs connected back to back preserves wideband voice by transferring the data between the two UXPADs in wideband format (IDP-WB).

• 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.

Opus transcoding options can be configured using the following parameters:

For more information, refer to Codec Entry - CLI.