This value allows you to control the time scaled play out of streaming audio from your multimedia record hardware to each individual phone line. This value allows the application to specify the audio adaptive "play out" buffering depth that will be used when the media engine sends locally recorded audio to each individual phone line. If your application does not configure the media engine to use locally recorded audio from your multimedia hardware, this value is ignored.

Under normal situations, this value can be set to 1 or 2. The lower the value, the less phone line transmit latency will be achieved. Setting this value higher will slightly increase audio transmit latency but will ensure continuous streaming transmit phone line audio. If for some reason you experience broken or "choppy" phone line transmit audio in your VOIP calls, increase this value until the transmit audio is clear and continuous. For multimedia record hardware and driver combinations that exhibit non real time timing characteristics, you may have to increase this parameter value to 4 or higher. Doing so will allow the media engine to properly compensate adaptively for the non real time nature of your selected record hardware/driver installation.

public int PhoneLinePlayoutBuffering

Public PhoneLinePlayoutBuffering As Integer

public:
int PhoneLinePlayoutBuffering

public int PhoneLinePlayoutBuffering