REST API Methods for this Resource
- 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 applicable for the SBC 1000/2000 platforms 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:
|OpusBandwidthSampleRate||No||No||Enum||4||Possible values: |
|VoiceModeBiteRate||Yes||No||Enum||1||Possible values: || Voice Mode Bitrate
Specifies if the decoder prefers the use of a constant bitrate versus a variable bitrate.
When cbr is set, the maximum average bitrate can still change, e.g., to adapt to changing
network conditions.This is applicable for Opus.
Valid values for the different Voice Mode Bite Rate:
|UseFEC||Yes||No||Enum||0||Possible values: ||Forward Error Correction. This FEC scheme adds redundant information about the previous packet (n-1) to the current output packet n. For each frame, the encoder decides whether to use FEC based on (1) an externally-provided estimate of the channel's packet loss rate; (2) an externally-provided estimate of the channel's capacity; (3) the sensitivity of the audio or speech signal to packet loss; (4) whether the receiving decoder has indicated it can take advantage of "in-band" FEC information. The decision to send "in-band" FEC information is entirely controlled by the encoder and therefore no special precautions for the payload or storage format have to be taken. This is applicable for Opus/Silk.|
|UseDTX||Yes||No||Enum||0||Possible values: ||Discontinuous Transmission. The Opus/Silk codec may operated with an adaptive bit rate. In that case, the bit rate will automatically be reduced for certain input signals like periods of silence. During continuous transmission the bit rate will be reduced, when the input signal allows to do so, but the transmission to the receiver itself will never be interrupted. Therefore, the received signal will maintain the same high level of quality over the full duration of a transmission while minimizing the average bit rate over time. In cases where the bit rate of Opus/Silk needs to be reduced even further or in cases where only constant bit rate is available, the Opus/Silk encoder may be set to use discontinuous transmission (DTX), where parts of the encoded signal that correspond to periods of silence in the input speech or audio signal are not transmitted to the receiver. On the receiving side, the non-transmitted parts will be handled by a frame loss concealment unit in the Opus/Silk decoder which generates a comfort noise signal to replace the non transmitted parts of the speech or audio signal. The DTX mode of Opus/Silk will have a slightly lower speech or audio quality than the continuous mode. Therefore, it is RECOMMENDED to use Opus/Silk|
|Complexity||No||No||int||0||Possible values: ||SILK offers 3 scalable complexity level for the encoder: 0 low, 1 medium, 2 high. Complexity can be scaled to optimize for CPU resources in real-time, mostly in trade-off to network bit rate. This is applicable for Silk.|