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
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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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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
--
Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: brian@freeswitch.com
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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
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
--
Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: brian@freeswitch.com
Mobile: 918-424-9378 <(918)%20424-9378>
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
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It is actually fairly common to want to stream audio/video to somewhere to process the data in real time, rather than waiting until recording is finished. 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<mailto: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<mailto: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<mailto: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 ? 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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. ----- Messaggio originale ----- Da: "Robert Mundkowsky" <rmundkowsky@ets.org> 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 It is actually fairly common to want to stream audio/video to somewhere to process the data in real time, rather than waiting until recording is finished. 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 >: <blockquote> 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: <blockquote> 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West | Co-founder and Developer Need Commercial support? email sales@freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian@freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com color-facebook-96.pngcolor-twitter-96.png _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org </blockquote> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org </blockquote> This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi thanks a lot for your answers. Do you split media stream ? I want to stream only caller voice for exemple. My ASR works better if it recognizes only one stream ;) thanks in advance 2018-03-01 17:24 GMT+01:00 Stefano Favaro <stefano.favaro@edistar.com>:
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.
------------------------------ *Da: *"Robert Mundkowsky" <rmundkowsky@ets.org> *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
It is actually fairly common to want to stream audio/video to somewhere to process the data in real time, rather than waiting until recording is finished.
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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Hi all, I worked on this POC script: https://github.com/Mickaelh51/rtp_parser/blob/master/pyshark_rtp_test.py It can parse pcap file with real sip calls. I can create a raw file with audio data (one file by leg), It's exactly I want do do ;) But now I wan to stream this to kaldi with websocket... it's hard :( (for me) Do you think I'm on the good way with this POC ? I can see other ways maybe: - https://freeswitch.org/confluence/display/FREESWITCH/mod_vlc - https://freeswitch.org/confluence/display/FREESWITCH/mod_shout Maybe it's easier to create a "gateway script" to read vlc or shout flow and send it to kaldi ? What do you think about this new idea ? The last chance: hack one of theses modules to put audio stream to kaldi... it's not the good way for my c++ skills ;) https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/asr... thanks in advance 2018-03-02 11:09 GMT+01:00 Mickael Hubert <mickael@winlux.fr>:
Hi thanks a lot for your answers. Do you split media stream ? I want to stream only caller voice for exemple. My ASR works better if it recognizes only one stream ;) thanks in advance
2018-03-01 17:24 GMT+01:00 Stefano Favaro <stefano.favaro@edistar.com>:
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.
------------------------------ *Da: *"Robert Mundkowsky" <rmundkowsky@ets.org> *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
It is actually fairly common to want to stream audio/video to somewhere to process the data in real time, rather than waiting until recording is finished.
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-bounc es@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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Hi, I must stream the voice to specific app (other server). This app processes the voice in live, not after end of call. 2018-02-28 19:47 GMT+01:00 Tihomir Culjaga <tculjaga@gmail.com>:
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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participants (5)
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Brian West
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Mickael Hubert
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Mundkowsky, Robert
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Stefano Favaro
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Tihomir Culjaga