I have a small application where I have created sip users in Freeswitch, now I want to know whether it is possible to have real-time transcription in FreeSwitch like Twilio media Streams API. Any help or resources would be appreciated.
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There is GitHub - alphacep/vosk-api: Offline speech recognition API for Android, iOS, Raspberry Pi and servers with Python, Java, C# and Node that has a third part module in their repo freeswitch/src/mod/asr_tts/mod_vosk at master · alphacep/freeswitch · GitHub