SIP audio settings

Setting

Description

Default  

Use Global SIP Station Audio Settings (Station Configuration Only)

This option specifies whether stations inherit the values that are defined for the global SIP station.

Audio Path

See the latest version of SIP Application Note on the Product Information site.

Dynamic

DTMF Type

The options are:

• Do not use RFC2833 inband only

• RFC2833 if supported, otherwise inband

• RFC2833 only

RFC2833 if supported, otherwise inband

DTMF Payload

This option sets the value that is used for the DTMF RTP payload type.

The acceptable values are 96-127. The vendor-specific values are: 100 and 102-105.

Note: The values 100, 102-105 should not be used for AudioCodes.

101

RTP DSCP Value

This option sets the Differentiated Services Code Point (DSCP) value of Quality of Service (QoS) in transmitted RTP packets.

The values are shown in both hex (00..3F) and related decimal (0..63) formats. Some values are also identified by the binary format, CS6.

The range of acceptable values is 00 (0, 000000) through 3F (63, 111111).  

Note: If this is a Polycom, SIP Soft Phone, or Interaction SIP Station managed IP phone, the default value is 2E (46, 101110) EF.

18 (24, 011000) CS53

Voice Activation Detection (VAD)

This option indicates whether Voice Activation Detection is enabled on your network. When Voice Activation Detection is used, no packets are sent for silence and bandwidth is saved. However, like compression, there is some loss of voice quality.

Not selected

Echo Cancellation

This option indicates whether echoes are removed from voice communications to improve the sound quality.

Selected

Allow Multiple Codecs in Outbound SDP Offer

This option indicates whether CIC delivers all of the available Codecs to the recipient endpoint when a user makes an outbound call. The recipient endpoint can then select which Codec it recognizes.

You set up Codecs in the Locations container.

Not selected