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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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To View Codec Entry

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

Call Routing - Codec Entry

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.

Call Routing - Codec Entry Highlighted

 

The Edit Selected Codec Entry window is displayed below.

Call Routing - Codec Entry Edit

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

Sonus recommends against editing the default codec entries. Edit only the user-defined codecs.

To Create Codec Entry

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

Call Routing - Codec Entry Fields

 

The Create New Codec Entry window is displayed.

Call Routing - Codec Entry Create Window

 

The following fields are displayed:

Codec Entry Parameters

Parameter Description
Name

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

Codec

Select the Codec from the drop-down 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

Max Average Bit Rate

The maximum Opus bit rate (in bits/second) used for the current session. The value ranges from 6000 to 510000 and the default value is 20000.

Use Cbr

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

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

Use Fec

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

Use Dtx

Use this parameter to specify whether or not to use Discontinuous Transmission (DTX). Applies to Opus only.
  • 0 – Do not use DTX
  • 1 –  Use DTX
Active Codec Set 

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
Fec Redundancy
Sets the level for Forward-Error-Correction (FEC) Redundancy [AMR only].  The default value of "0" means FEC redundancy is disabled.
Coding Rate

This parameter is used to set the corresponding coding rate for G7221 codec. The values are:

  • 16
  • 24
  • 32

This parameter is enabled only for G7221 codec.

Mode Change Neighbor

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
Packet Size

The packet size in milliseconds (ms). Options are based on the type of codec chosen. 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)

Preferred RTP Payload Type

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).

Silence Suppression

Enable/disable Silence Suppression mode.

  • disable (default)
  • enable
Display LevelTo display different levels of output information in show commands.

Law

Specify the G711 law to use, values are:

  • A Law
  • U Law
  • derivedFromOtherLeg (default)
Max Interleave Depth

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.

This field only applies to EVRC and EVRCB calls.

The parameter, Max Interleave Depth, is visible only when:

  • The Codec parameter is set to evrc and evrcb .
  • The Packet Size parameter is set to 40 or 60.
DTMF
Relay

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.

Remove DigitsEnables the removal of DTMF digits from the media stream. This applies only if DTMF relay is configured as Out-of-Band or RFC 2833. The default setting is Enabled.
Fax

Failure
Handing

Specifies the behavior when a fax tone is detected but the treatment fails for any reason. The behavior can be:
• Disconnect—Release the call.
• Continue—Continue to process the call.

The default setting is Continue.

Tone Treatment

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.

For G.711 calls, Notify Peer, Disconnect, Fax Relay, and Fax Relay or Fallback to G.711 require allocation of a compression resource.

Modem

Failure
Handing

Specifies the behavior when a modem tone is detected but the treatment fails for any reason. The behavior can be:
• Disconnect—Release the call.
• Continue—Continue to process the call.

The default setting is Continue.

Tone Treatment

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.

applyFaxTreatment is not supported for Gateway Links.

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.

Call Routing - Codec Entry Highlighted

 

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

Call Routing - Codec Entry Fields

 

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

Call Routing - Codec Entry Copy Window

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.

Sonus recommends against deleting the default codec entries.

Call Routing - Codec Entry Highlighted

 

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

Call Routing - Codec Entry Delete Confirmation

 

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

 

 

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