simvoice - SIP Voice call/Instant messaging converter (server)
simvoice -a < sphinx argument file > [options]
The binary and sources are available. The code was tested only on Linux but should be able to run on other Unix platforms.
simvoice is a translator between SIP based audio call and SIP based instant messaging (IM). It allows a audio-based phone user to interact with a text-based IM user, via text-to-speech and speech-to-text conversion.
- supports SIP version 2.0 (RFC 2543, RFC 3261)
- supports only G.711 Mu-Law audio
- does silence detection to find the end of a talk-spurt
- uses RTP (RFC 1889, RFC 1890) in streaming
- uses IBM ViaVoice TTS as text-to-speech engine
- uses CMU Sphinx2 as speech-to-text engine
- CMU Sphinx2 should be downloaded and installed properly on the machine where simvoice will be run. Sphinx2 ASR engine used by simvoice needs a long list of arguments to be initialized, which give the engine the information such as the location of the dictionary, input model database and other configuration parameters. Current version of simvoice takes a list of arguments as a text file and the file name should be specified as an argument when simvoice is initiated. See example sphinx.arg. More detail about arguments, see Sphinx2 User Guide.
- IBM ViaVoice TTS should be downloaded and installed properly on the machine where simvoice will be run. The initialization file of the engine eci.ini that comes with installation should be in the simvoice directory.
- -v
- Print version information and exit.
- -h
- Print usage and exit.
- -d category
- Makes simvoice print out debugging information for the specified category. Current supported categories include all, sql, net, sdp, misc.
- -p port
- Use specified port number instead of default 5060 for SIP request waiting.
- -t value
- Use specified value as the threshold value for silence detection.
$ simvoice -a sphinx.arg -p 6000 -d misc
will initiate simvoice with sphinx argument file [sphinx.arg], waiting port 6000 and debugging information in category [misc]
simvoice was only tested with sipc as a voice caller and an IM user, which uses external tool "ratmedia" in streaming.
sipc
Naoya Seta, at Department of Electrical Engineering, Columbia University
Prof. Henning Schulzrinne
Kundan Singh
Xiaotao Wu
at Columbia University, Department of Computer Science
Copyright 1998-1999 by Columbia University; all rights reserved
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