Specifies the size of the RTP packet in samples.
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Value |
Integer: 1–4000 samples. Typical settings are 160 (20 ms) or 240 (30 ms). DEFAULT: 240 |
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How to set |
If using the Management Station, set on the Speech Server service. If not using Management Station, set in an NSS configuration file (user-NSSxx.txt). |
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Usage |
Seldom changed. |
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Performance |
Increasing the packet size reduces the overhead per packet by adding more audio to each packet. It also reduces the frequency that Vocalizer polls for audio, which reduces CPU requirements, but adds a slight latency. A value of 240 translates to 30 milliseconds. This is the optimal setting for Nuance Speech Server. Nuance recommends sending packets to Speech Server at the same rate. If your system sends at a different rate, there are no negative effects in the runtime system but any created WCR (whole call recording) audio files may have small degradations. |
Use this parameter to improve text-to-speech quality.
Several TTS parameters affect the RTP sending speed of audio samples from Speech Server to the telephony interface. When the speed is too slow, accelerates audio delivery for a period of time and then resumes normal speeds (as determined by the sampling rate).
Speech Server detects delays by comparing the number of “sent ahead” samples to the value of rtpLowerBoundarySamples. If the number of samples is less than the value of the parameter, Speech Server accelerates the delivery speed.
Gateways typically set this parameter to 20 ms or 30 ms (160 = 20 ms, 240 = 30 ms). Nuance recommends that you align settings with the gateway configuration.
Note: Nuance Speech Server automatically adjusts packet sample size to match the value in a ptime (packet size) value requested by the gateway in the SIP INVITE.
Supported for MRCPv2 browsers. (Ignored for MRCPv1 browsers.)
Related parameters
