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The Codec Entry screen enables you to define entries for audio encoding methods and their associated attributes. The parameters available change, depending on which codec you select for audio encoding.

For a list of supported audio codecs, refer to Supported Codecs and Transcoding..

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For a list of supported audio codecs, see Audio Codecs page.

To View Codec Entry

On SBC main screen, go to Configuration > System Provisioning > Category: Call Routing > Codec Entry. The Codec Entry window is displayed.

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To Edit Codec Entry

To edit any of the Codec Entries in the list, click the radio button next to the specific Codec Entry name.

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The Edit Selected Codec Entry window is displayed below.

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Make the required changes and click Save at the right hand bottom of the panel to save the changes made.

To Create Codec Entry

To create a new Codec Entry, click the New Codec Entry tab on the Codec Entry List panel.

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The Create New Codec Entry window is displayed.

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The following fields are displayed:

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The codec entry ID used to identify a particular codec entry.

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Select the Codec from the drop-down list (refer to Audio Codecs for the codec list):

• amrBandwidthEfficient
• amrCrc
• amrCrcInterleaving
• amrCrcRobustSorting  
• amrInterleaving
• amrInterleavingRobustSorting
• amrIuUP
• amrOctetAligned  
• amrRobustSorting
• amrwbBandwithEfficient
• amrwbCrc
• amrwbCrcInterleaving
• amrwbCrcRobustSorting
• amrwbInterleaving
• amrwbInterleavingRobustSorting
• amrwbOctetAligned
• bv16
• bv32
• bv32Fec
• efr
• evrc
• evrc0
• evrc1
• evrc1Fr
• evrcb
• evrcb0
• evrcb1
• evrcb1Fr
• g711 (default)
• g711ss
• g722
• g7221
• g7221ss
• g7231
• g7231a  
• g726
• g726ss  
• g728
• g728ss
• g7291
• g729a
• g729ab
• gsm
  
• ilbc  
• ilbcss
• isac
• I16-16
• msrta16
• msrta8
• opus
  
• silk12
• silk16
• silk24
• silk8
• speex16
• speex16Fec
• speex32
• speex8
• speex8Fec
• amrwbCrcInterleavingRobustSorting

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Max Average Bit Rate

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Use Cbr

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Use this parameter to specify variable or constant bit rate. Applies to Opus only.

• 0 – Use variable bit rate
• 1 –  Use constant bit rate

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Use Fec

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Use this parameter to specify whether or not to use Forward Error Correction (FEC). Applies to Opus only.

• 0 – Do not use FEC
• 1 –  Use FEC

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Use Dtx

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• 0 – Do not use DTX
• 1 –  Use DTX

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The active code set is applicable to certain AMR narrow-band codecs. Multiple rates may be selected using comma (,). Valid values are:

• AMR-0-4.75kbps
• AMR-1-5.15kbps
• AMR-2-5.90kbps
• AMR-3-6.70kbps
• AMR-4-7.40kbps
• AMR-5-7.950kbps
• AMR-6-10.20kbps
• AMR-7-12.20kbps

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This parameter is used to set the corresponding coding rate for G7221 codec (refer to Audio Codecs for the codec list).

The values are:

• 16
• 24
• 32

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Enable flag on peer or route PSP to cause SBC to configure DSPs to force mode change to neighboring modes in active codec set as per RFC4867 (applies to AMR and AMRWB).

• disable (default)
• enable

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Use this flag to determine the initial codec mode of an AMR/AMR-WB transcoded call once the call is established.

• disable (default) – AMR/AMR-WB call starts with the highest rate in the active mode set.
• enable – AMR/AMR-WB call starts with a rate determined by the following algorithm:

1. 1. If one codec mode is included in the mode-set, it is the initial codec mode.
   2. If two or three codec modes are included in the mode-set, the initial codec mode is the codec mode with the lowest rate.
   3. If four or more codec modes are included in the mode-set, the initial codec mode is the codec mode with the second lowest rate.

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The packet size in milliseconds (ms). Options are based on the type of codec chosen (refer to Audio Codecs for the codec list).

Example packet sizes:

• g711, g711ss : 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60 (default = 10)
• g7221, g726 : 10, 20, 30, 40 (default = 10)
  
• g723, g723a : 30, 60, 90, 120, 150 (default = 30)
  
• g729a, g729ab : 10, 20, 30, 40, 50, 60   (default = 10)

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Preferred RTP Payload Type

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Specifies the preferred Real Time Protocol (RTP) payload type to be included in the RTP header of the data packet. For audio codecs G726, G723, and G729 the value is a range 0-128. The value for the preferred RTP payload type is fixed at 128 for all other audio encoding methods except iLBC and iLBC with Silence Suppression. For iLBC and iLBC with Silence Suppression, the preferred RTP payload type can be set to any value in the range of 0 to 127 (with no default value).

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Enable/disable Silence Suppression mode.

• disable (default)
• enable

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Law

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Specify the G711 law to use, values are:

• A Law
• U Law
• Derive From Other Leg (default)

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This parameter specifies the amount of interleaving an endpoint can deal with. The value ranges from 0 to 7 and the default value is 0.

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The possible values are:

• None—Leaves the DTMF tones in-band as encoded audio.
• Out-Of-Band—Carries DTMF in the signaling protocol.
• RFC 2833—Encodes DTMF into RTP using a format and payload type distinct from the audio encoding.
• Either OOB or 2833—Out-of-Band and RFC 2833 are equally received and only one is transmitted. The one transmitted is the one preferred by the peer or RFC 2833 as the default.
• Both OOB And 2833—Out-of-Band and RFC 2883 are equally received and both can be transmitted. This option would normally be used only in the case where the OOB DTMF signaling is absorbed and not regenerated. For example, the OOB DTMF might go to an application server that needs to detect the DTMF for control purposes but does not process RTP and the 2833 DTMF would go to the destination media address as part of the RTP stream.

The default setting is None.

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Failure
Handing

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Tone Treatment

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Specifies the treatment taken when the fax tone is detected, which can be:
• None—Do nothing when the fax tone is detected.
• Notify Peer—For SIP signaling, notify the peer when the fax tone is detected and let the peer decide the next action.
• Disconnect—Disconnect the call when the fax tone is detected.
• Fallback to G.711—Fall back to G.711 when the fax tone is detected.
• Fax Relay—Switch to fax relay (T.38) when the fax tone is detected.
• Fax Relay or Fallback to G.711—Switch to fax relay (T.38) if supported or fall back to G.711 when fax tone is detected.
• Ignore Detect Allow Peer to Negotiate Fax Relay—Accept a T.38 reINVITE (either from a calling party or a called party) without detecting the fax tone.
The default setting is None.

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Failure
Handing

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Tone Treatment

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Specifies the treatment taken when the modem tone is detected, which can be:
• None—Do nothing when the modem tone is detected.
• Notify Peer—Notify the peer when the modem tone is detected.
• Disconnect—Disconnect the call when the modem tone is detected.
• Fallback to G.711—Fall back to G.711 when the modem tone is detected.
• Apply Fax Treatment—Treat the modem tone as a fax tone, and apply the fax treatment for the selected codec.

The default setting is None.

To Copy Codec Entry

To copy a Codec Entry and optionally make changes to the copy, click the radio button next to the specific Codec Entry to highlight the row.

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Click Copy Codec Entry tab on the Codec Entry List panel.

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The Copy Selected Codec Entry window is displayed along with the field details which can be edited.

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Make the required changes to the required fields and click Save to save the changes. The copied Codec Entry is displayed at the bottom of the original Codec Entry in the Codec Entry List panel.

To Delete Codec Entry

To delete any Codec Entry, click the radio button next to the specific Codec Entry which you want to delete.

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Click Delete at the end of the highlighted row. A delete confirmation message appears seeking your decision.

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Click OK to remove the specific Codec Entry from the list.

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