I've worked on this some more, and have been able to watch a little more of what's happening by splitting into two FreeSWITCH instances and sending traffic between them.

I can make the jitter buffer and delay_echo provide the required delay, and this works fine with a basic SIP softphone.

What's happening is that mod_portaudio is providing L16 at high sample rates and isn't transcoding it even across servers.

It seems that I need to convert L16 48K to (say) PCMA or OPUS or something that delay_echo or the jitterbuffer could deal with.

Is that something that's possible? I know I can eavesdrop or send to a conference, but is there a way I can easily convert from portaudio to something more generic to send to the dialplan?

Erik

On Sun, Jan 28, 2018 at 8:36 PM, Alexey Sibyakin <alex@freeswitch.com> wrote:
<action application="set" data="rtp_jitter_buffer_during_bridge=true"/>
<action application="export" data="nolocal:jitterbuffer_msec=3000:3200:20"/>

here you need to choose Alice or Bob will have JB, not both of them only half

On Fri, Jan 26, 2018 at 12:35 PM, Erik M. Devane - Comms Guy <emdevane@gmail.com> wrote:
I'm sorry, I'm just not having any luck getting this going.

Ideally, I'd like to be able to originate the portaudio endpoint into park, and eavesdrop on it there, but I understand that's convoluted to work with, so as a test I have a single extension:

<extension name="portaudio_USB">
    <condition field="destination_number" expression="^1(\d{2})$">
    <action application="answer"/>
<action application="set" data="rtp_jitter_buffer_during_bridge=true"/>
<action application="export" data="nolocal:jitterbuffer_msec=3000:3200:20"/>
  <action application="bridge" data="portaudio/endpoints/USB_01"/>
    </condition>
  </extension> 

Dialing 101 from a SIP client (MicroSIP) as user 1000 from the default configuration causes me to be bridged to the portaudio endpoint (I can hear input from the channel), but there's no delay at all.





On Thu, Jan 25, 2018 at 10:13 AM, Brian West <brian@freeswitch.com> wrote:
also you'll have to set rtp_jitter_buffer_during_bridge to true.

/b


On Thu, Jan 25, 2018 at 2:22 AM, Alexey Sibyakin <alex@freeswitch.com> wrote:
Then you should try to set JB for b-leg via export nolocal.

On Thu, Jan 25, 2018 at 3:44 PM, Erik M. Devane - Comms Guy <emdevane@gmail.com> wrote:
Thank you for the reply. Here's what I tried:

<action application="set" data="jitterbuffer_msec=3000:3200:20"/>

Works for calls from SIP clients but not for PortAudio calls.

Same with delay_echo - the CODEC is seen as RAW and so the jitterbuffer doesn't seem to be engaged (read_frame->packet doesn't have anything to work with so it skips the buffer).

I have tried to have PortAudio > extension 1 > bridge to extension 2, which produces better results, but (in my hands at least) it causes memory leaks and is unwieldy.




On Wed, Jan 24, 2018 at 11:59 PM, Alexey Sibyakin <alex@freeswitch.com> wrote:
JB should be fine. Any details of unsuccess?

On Thu, Jan 25, 2018 at 2:45 PM, Erik M. Devane - Comms Guy <emdevane@gmail.com> wrote:
I've been working on using FreeSWITCH as a bridge between different devices, and I need to delay all calls by 2-3 seconds in order for the external systems to keep up.

This delay should happen at the start of a call, and all subsequent audio should sound fine, just late.
I know that doing this sounds silly - I'm sorry about that.

Here are my approaches:

1) Eavesdrop on a call send through delay_echo - this works for SIP clients but I have to do an absolute_codec_string as it seems that CODEC support is limited, and I'm often getting no delay at all with my PortAudio clients.

2) Use lua to do a session freeswitch.msleep on answer - exactly what Confluence tells me to avoid - if I sleep but don't consume audio, I think I can get the desired result.

3) Mess with the jitterbuffer - I haven't had any success with this yet.

4) Increase the buffer in mod_PortAudio.

Writing this out makes me sure that I'm missing something - a way of getting delay_echo to work for all calls, or a way to pause session media to get the desired effect.

Can you please help?

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Alex Sibyakin | Support Engineer
Need commercial support? Contact sales@freeswitch.com for details.

_________________________________________________________________________
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Official FreeSWITCH Sites
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
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--

Alex Sibyakin | Support Engineer
Need commercial support? Contact sales@freeswitch.com for details.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
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_________________________________________________________________________
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
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_________________________________________________________________________
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
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--

Alex Sibyakin | Support Engineer
Need commercial support? Contact sales@freeswitch.com for details.

_________________________________________________________________________
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

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