Troubleshooting with parameters

Parameters useful for troubleshooting are described below. You will not normally need to modify these parameters; you will most likely use them in conjunction with Nuance Technical Support.

Analyzing SIP Integration

Voice platform integrators can use this parameter when developing a system:

Parameter	Description	Value
server.mrcp2.sip.logLevel	Specifies the logging level of the SIP stack.	Integer: -1 to 8 (9 and 666 for Nuance internal use) DEFAULT: -1 (no SIP logging)

Parameter

Description

Value

server.mrcp2.sip.logLevel

Specifies the logging level of the SIP stack.

Integer: -1 to 8 (9 and 666 for Nuance internal use)

DEFAULT: -1 (no SIP logging)

Improving text-to-speech quality

These parameters are useful for tuning Vocalizer:

Parameter	Description	Value
server.mrcp2.rsspeechsynth.audioThreadNumber	Specifies the number of sending threads.	Integer DEFAULT: 20
server.mrcp2.rsspeechsynth.rtpBufferFillMultiplier	Adjusts the RTP sending speed by filling packets more quickly.	Integer: 1–10. DEFAULT: 2
server.mrcp2.rsspeechsynth.rtpLowerBoundarySamples	Adjusts the RTP sending speed by setting the minimum number of samples that can be sent ahead.	Integer: 0–rtpUpperBoundarySamples samples. DEFAULT: 300
server.mrcp2.rsspeechsynth.rtpPacketSamples	Specifies the size of the RTP packet in samples.	Integer: 1–4000 samples. Typical settings are 160 (20 ms) or 240 (30 ms). DEFAULT: 240
server.mrcp2.rsspeechsynth.rtpSendRate	Specifies the number of audio samples sent per second.	Integer: 1–INT_MAX samples. DEFAULT: 8000
server.mrcp2.rsspeechsynth.rtpUpperBoundarySamples	Adjusts the RTP sending speed by setting the maximum number of samples that can be sent ahead.	Integer: rtpLowerBoundarySamples–INT_MAX samples. DEFAULT: 600

Managing text-to-speech performance

When Speech Server detects delays in sending audio samples from Speech Server to the MRCP client, it accelerates audio delivery for a period of time and then resumes normal speeds (as determined by the sampling rate). Several text-to-speech (TTS) parameters affect the RTP sending speed:

Speech Server detects delays by comparing the number of “sent ahead” samples to the value of server.mrcp1.rsspeechsynth.rtpLowerBoundarySamples or server.mrcp2.rsspeechsynth.rtpLowerBoundarySamples. If the number of samples is less than the value of the parameter, Speech Server accelerates the delivery speed.
Speech Server uses the value of server.mrcp2.rsspeechsynth.rtpBufferFillMultiplier or server.mrcp1.rsspeechsynth.rtpBufferFillMultiplier to generate and send samples more quickly. If the parameter is set to 0, the samples are generated and sent as fast as possible without artificial pauses between sent packets.
Speech Server continues the accelerated delivery until the number of “sent ahead” samples is greater than the value of server.mrcp1.rsspeechsynth.rtpBufferFillMultiplier (at which time, Speech Server resumes sending TTS audio data at normal speeds.

Increasing data written to diagnostic logs

This parameter controls the data written to diagnostic logs. To help troubleshooting, Nuance technical support might request that system administrators set this parameter:

Parameter	Description	Value
swirec_diag_tags_enable	Controls the logged output by enabling tags in diagnostic log files.	Integers representing one or more tags in the TRC tagmap files. DEFAULT: (none)

Parameter

Description

Value

swirec_diag_tags_enable

Controls the logged output by enabling tags in diagnostic log files.

Integers representing one or more tags in the TRC tagmap files.

DEFAULT: (none)

Reducing network latencies

Optional. Use these parameters to define rules for fetching documents through a proxy server. The advantage of these parameters is that they apply to fetches by Speech Server, Recognizer, and Vocalizer instead of configuring proxies for each individual engine.