Hello, I'm currently using:
Freeswitch with mod_unimrcp and the plugin from UniMRCP (http://unimrcp.org/gsr)
It connects to Google Speech API for audio transcription in realtime using mrcp.
Stefano.
A: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org>
Inviato: Mercoledì, 28 febbraio 2018 20:17:01
Oggetto: Re: [Freeswitch-users] Send RTP to external server
For example, if want to convert audio to text (ASR) then you do not want to wait until the conference is over before you start the ASR. Imagine, your ASR takes 1 minute to convert 1 minute of audio then if a conference is 30 minutes long, you would have to wait another 30 minutes for the ASR to finish before you could do something with the text.
I believe you can use gstreamer (https://gstreamer.freedesktop.org/) to handle receiving the RTP from a FreeSWITCH conference. I think it allows you to sample frames if you want to get frames from video granted I think you are only interested in audio, but I think you still have to develop a daemon that understand SIP/RTP in order to talk to FreeSWITCH. Another approach might be to create a plugin for the Unimrcp (MRCPv2 server http://www.unimrcp.org/ ) and talk to the FreeSWITCH using their mrcp module and tight that to the conference call. You might be able to hack the Kaldi plugin for Unimrcp to do what you want.
Robert
From: FreeSWITCH-users [mailto:freeswitch-users-bounces@lists.freeswitch.org]
On Behalf Of Tihomir Culjaga
Sent: Wednesday, February 28, 2018 1:47 PM
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Send RTP to external server
why do you want to stream... isn't it enough just to rsync the file once the call is finished ?
On 28 February 2018 at 16:16, Mickael Hubert <mickael@winlux.fr> wrote:
Hi
thanks a lot for your answer.But I want to send stream to external server, not record in audio file. i can use record to "capture" the voice, but to stream it, it's more complicated ;)
thanks in advance
2018-02-21 0:50 GMT+01:00 Brian West <brian@freeswitch.com>:
you can already do this without SIPREC in freeswitch. By setting the RECORD_READ_ONLY or RECORD_WRITE_ONLY variables.
/b
On Fri, Feb 16, 2018 at 10:33 AM, Mickael Hubert <mickael@winlux.fr> wrote:
Hi list,
I want to record each call through freeswitch. But i want record only caller (SSRC 1) OR callee (SSRC 2) voice (not both).
I read about SIPREC, Jack, etc ... not interesting
Do you have a idea for me please ?
Thanks in advance
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