To create or modify a Voice Codec Profile:
- In the WebUI, click the Settings tab.
- In the left navigation pane, go to Media > Media Profiles.
- From the Create Media Profile drop down box, select Voice Codec Profile.
Modifying a Voice Codec Profile
- Click the expand ( ) Icon next to the entry you wish to modify.
- Edit the entry properties as required, see details below.
Creating a Voice Codec Profile
- From the Create Media Profile drop-down menu at the top of the Media Profiles page, select Voice Codec Profile.
- Complete Voice Codec Configuration fields (see Field Definitions below), and click OK.
Voice Codec Profiles - Field Definitions
Specifies the voice codec 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). The choice for codec will depend on the interoperability requirements for connecting to other voice peers, as well as on the bandwidth requirements. Most codecs use data compression algorithms, which saves the bandwidth, but it slightly reduces the voice quality, whereas G.711 does not use compression and therefore requires most bandwidth.
Note: The G.722.2 Codec only appears in the Codec drop-down menu if the AMR license is installed.
- G.722 Wideband Codec: this codec supports only 20ms packet sized and a bit-rate of 64Kbps.
- There is no fax data tone detection support in the G.772 or AMR-WB channels.
The following codecs are supported on SBC 1000/2000 platform:
* SBC 1000/2000 supports 32 Kbps bit rate only.
** Both G.729A and G.729AB are supported. The variant used when selecting "G.729" from the Codec field depends upon the setting of Media List flag "Silence Suppression". If Silence Suppression is enabled, G.729AB is used.
Specifies the 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.
Specifies (indirectly) the number of bytes in a single packet.
The payload size is a multiple of the codec sample size. For example: the codec sample interval for G.711u/a is 10 ms. Therefore, the voice payload sizes are 10, 20, 30, 40, 50, 60. The codec sample interval for G.723.1 is 30 ms and therefore the payload sizes are 30, 60, 90.
Smaller payload sizes decrease audio transport latency at the expense of higher bandwidth consumption.
Larger packet sizes reduce the use of bandwidth. The larger the payload size, the fewer and larger the packets. When larger payload sizes are used, fewer L2/Ethernet headers as well as fewer IP/UDP/RTP headers are used. The disadvantage is that if UDP packets get lost, the impact on the voice quality will be much higher because a single packet contains more raw voice data.
Total packet size = (L2 header: MP or FRF.12 or Ethernet) + (IP/UDP/RTP header) + (voice payload size)
Specifies the payload type for this profile. Only available when either G722.2 and G.726 is the specified codec.
Specifies the payload format for this profile. This field is only available when the G.722.2 codec is selected.
- Bandwidth-Efficient Mode
- Octet-Aligned Mode
For a given session, the payload format can be either bandwidth efficient or octet aligned, depending on the mode of operation that is established for the session via out-of-band means. In the octet-aligned format, all the fields in a payload, including payload header, table of contents entries, and speech frames themselves, are individually aligned to octet boundaries to make implementations efficient. In the bandwidth-efficient format, only the full payload is octet aligned, so fewer padding bits are added.