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To create or modify a Voice Codec Profile:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, go to Media > Media Profiles.
  3. From the Create Media Profile drop down box, select Voice Codec Profile.

Modifying a Voice Codec Profile

  1. Click the expand () Icon next to the entry you wish to modify.
  2. Edit the entry properties as required, see details below.

Creating a Voice Codec Profile

  1. From the Create Media Profile drop down box at the top of the Media Profiles page, select Voice Codec Profile.

  2. From the Codec drop down box, select a codec.

  3. Configure the files and click OK. See field definitions below.

    Create Voice Codec Profile

     

Voice Codec Profiles - Field Definitions

Configure Codecs

ParameterDescription
Description A name/description for the codec.
Codec

Specifies the voice codec and encoding scheme used towards the IP side of a VoIP call.

The selected codec affects the audio quality and bandwidth consumption of VoIP calls to which you apply the Voice Codec Profile (in the Media Profiles List). The codec choice depends on the interoperability requirements for connecting to other voice peers and bandwidth requirements. Most codecs use data compression algorithms, which saves bandwidth but slightly reduces the voice quality. G.711 does not use compression and requires more bandwidth.

The following codecs are supported on 

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platform:

  • G.711 A-Law
  • G.711 u-Law
  • G.722
  • G.722.2
  • G.723.1
  • G.726
  • G.729
  • Opus
  • SILK (see SILK Codec for more information)

Codec - Additional Specifications

  • G.722 Wideband Codec: Supports 20ms packet sized and a bit-rate of 64Kbps only.
  • G.722 or AMR-WB channels: Does not support fax data detection support.
  • G.722.2: Appears only if the AMR License is installed.
  • G.726: Supports 32 Kbps bit rate only.
  • G.729: Supports G.729A and G.729AB. The variant used when selecting G.729 depends upon the setting of Media List flag Silence Suppression. If Silence Suppression is enabled, G.729AB is used.
  • Opus: Supported in SBC SWe Lite only.
  • SILK: Supported in the
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    only.

NOTE: Applicable to SWe Lite initiating FAX setup. A G.711 codec must be used for fax detection; tones are detected on the G711 egress (answering CED tone) leg. Otherwise the peer has to initiate fax path.

 

Bandwidth

Specifies the bandwidth sampling frequency. This parameter has the following options:

  • Narrowband - The 
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    only uses the narrowband mode either to interface to PSTN networks, or on low-end devices that support 8000 Hz or less sampling frequency.
  • Wideband (default) - The 
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    uses the wideband mode for all IP platforms that support 16,000 Hz or less sampling frequency.

This parameter applies only to SILK.

Rate

Specifies the voice sampling rate in bits/sec used by the codec.

This parameter applies to G.722.2, G.723.1, G.729, Opus, and SILK codecs only. For all other codecs, the voice sampling rate is fixed and defined in the appropriate specification for that codec.

Payload Size

The recommended length of time in milliseconds rounded up to the next full integer value represented by the media in a packet. Valid entry: 10, 20, 30, 40, 60, 80, 90 (specific codec determines values available).

  • Smaller payload sizes decrease audio transport latency at the expense of higher bandwidth consumption.
  • Larger packet sizes reduce the bandwidth. The larger the payload size, the fewer and larger the packets. With larger payload sizes, fewer L2/Ethernet and IP/UDP/RTP headers are used; a disadvantage is that if UDP packets get lost, the impact on the voice quality is higher because a single packet contains more raw voice data.

SILK supports only the 20, 40, and 80 payload sizes.

For Re-Invite Only:

If the SBC receives a larger than configured payload size from the peer offer in the re-invite, the SBC rejects with a 488 'Not Acceptable Here' response. The call rolls back to the previous negotiated offer answer.

Payload Type

Specifies the payload type for this profile. Applies to G722.2, G.726, Opus, and SILK codecs only.

For SILK and Opus, the Payload Type option is 96 - 127. The default value is 120.

Payload Format

Specifies the mode for the payload: Bandwidth Efficient Mode or Octet Aligned Mode. Default entry: Bandwidth Efficient Mode. This setting applies to G.722.2 only.

Voice Bit Rate

The maximum Opus bit rate (in bits/second) used for the current session. Valid entry: 0 (use variable bit rate) or 1 (use constant bit rate). Default entry: CBR.

Parameter applies to Opus only.

Use FEC

Specifies whether to use Forward Error Correction (FEC).

Valid entry for Opus: 0 (do not use FEC) or 1 (use FEC). Default entry for Opus: 0.

Valid entry for SILK: False (do not use FEC) or True (use FEC). Default entry for SILK: False.

Parameter applies to Opus and SILK only.

Use DTX

Specifies whether to use Discontinuous Transmission (DTX).

Valid entry for Opus: 0 (do not use DTX) or 1 (use DTX). Default entry for Opus: 0.

Valid entry for SILK: False (do not use DTX) or True (use DTX). Default entry for SILK: False.

Parameter applies to Opus and SILK only.

Complexity Level

Scales the complexity to optimize for CPU resources in real-time, which is mostly in trade-off to network bit rate. The following options are available for this parameter: 0, 1, or 2. Default entry: 0.

This parameter applies only to SILK.

 

SILK Codec

The 

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supports the SILK audio codec. Skype designates SILK as an internet wideband audio codec for use in VoIP. SILK operates at two different sampling rates: 8000 Hz narrowband and 16,000 Hz wideband (see the SILK Bandwith Options table). These rates allow for the capture of higher frequencies, which provide fuller sound, while also allowing interoperability with the Public Switched Telephone Network (PSTN). SILK has Low Bit Rate Redundancy (LBRR), also called Forward Error Correction (FEC), which protects the 
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against packet loss.

The network bit rate of SILK is adaptive within the range that the following table specifies. The 

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defines and modifies the average network bit rate in real-time, while the actual bit rate depends on the input signal and change over time. The bit rate can dynamically change within that range. Since all other parameters are equal, the higher bit rates result in higher audio quality.

SILK Bandwidth Options

Audio BandwidthFrequency (Hz)Bit Rate (KBPS)Description

Narrowband

80006 - 20The 
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only uses the narrowband mode either to interface to PSTN networks, or on low-end devices that support 8000 Hz or less sampling frequency.

Wideband

16,0008 - 30The 
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uses the wideband mode for all IP platforms that support 16,000 Hz or less sampling frequency.
The following table outlines the SBC 1000/2000 SILK performance capacity.

SBC 1K/2K SILK Performance Capacity

Configuration (post-Release 6.1.x)Example SKU(s)Number of DSPsNumber of Sessions, SILK NB <-> G.711

SBC 1000, SIP entry model

SBC-1K-R-SIP-E

114

SBC 1000, SIP

SBC-1K-R-SIP

342

SBC 1000 with PRI

SBC-1K-R-FXS8FXO-P, SBC-1K-R-P

228

SBC 1000 with FXx, no PRI

SBC-1K-R-FXSFXO, SBC-1K-R-FXS

114

SBC 1000 gateways

SBC-1K-R-4P-FXSFXO-GW114

SBC 2000, 1 DSP

SBC-2K-R-1

160

SBC 2000, 2 DSP

SBC-2K-R-2

2120

SBC 2000, 4 DSP

SBC-2K-R-4

4240

SBC 2000, 6 DSP

SBC-2K-R-6

6360

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