- GET voicecodecprofile
- GET voicecodecprofile id
- POST voicecodecprofile id
- PUT voicecodecprofile id
- DELETE voicecodecprofile id
|Parameter Name||Required||Service Affecting||Data Type||Default Value||Possible Values||Description|
|Description||No||No||string||none||64 - Max Length||Identifies this profile so it can be easily recognized when selecting a codec.|
|MediaType||Yes||Yes||Enum||3||Possible values: ||
Specifies the voice coding and encoding scheme used towards the IP side of a VoIP call.
The chosen codec affects the audio quality and bandwidth consumption of VoIP calls to which you apply this Voice Codec Profile (in the Media List Profile).
Only the following codecs are currently supported:
|VoiceRateInBitsPerSecond||No||No||int||0||Possible values: ||
Voice sampling rate in bits/sec to be used by the codec. This setting applies to G.723.1 and G.726 codecs only.
For all other codecs, the voice sampling rate is fixed and defined in the appropriate specification for that codec.
This option is available when Codec is set to G.723.1, G.726, G.722, or G.722.2.
|PayloadType||No||No||int||0||Possible values: ||
Specifies the payload type for this profile. Acceptable values for different codecs are as follows:
This option is available when Codec is G.726 or G.722.2.
The Payload Type selected for G.726 or G.722.2 must not conflict with that selected for Digit Relay (in Media List Profile).
|PTimeInMilliSeconds||No||No||int||0||Possible values: ||Real-Time Transport Protocol (RTP) packet payload size in milliseconds. Smaller payload sizes decrease audio transport latency at the expense of higher bandwidth consumption.
Valid values for the different media types are:
|PayloadFormat||Yes||Yes||Enum||0||Possible values: ||Payload format for packetization of AMR and AMR-WB encoded speech signals into the Real-time Transport Protocol.
Following payload formats are supported: