Recordings; Boosting SIP/phone quality or volume

[Ally]

Hi Community (Fred),

I have not looked at the source code yet for this, but is it possible to boost or enhance the audio of SIP when the recordings are stitched together?

The audio from a PC mic (webrtc audio) is great during session and in the recording. However, when a participant joins by SIP, the final recording shows a huge contrast in audio levels. I tested myself to make sure speaking distance from mic was the same in both cases.

Extra info: 

We have control over the trunking and asterisk server which has an custom extension to freeswitch. We couldn't see any issues with a drop in quality on our side. 

Everything is localised to the same geolocation and datacentre.

Thanks.

Ally.

[Fred Dixon]

Hi Allly,

> The audio from a PC mic (webrtc audio) is great during session and in the recording. However, when a participant joins by SIP, the final recording shows a huge contrast in audio levels. I tested myself to make sure speaking distance from mic was the same in both cases.

The great thing about WebRTC audio is that it has auto levelling, so it automatically increases the gain for the participants.

> I have not looked at the source code yet for this, but is it possible to boost or enhance the audio of SIP when the recordings are stitched together?

Actually, there is only one WAV file for the meeting (whereas each user has their own webcam file if shared).  FreeSWITCH is writing the WAV file based on the audio it receives.

FreeSWITCH does give users the ability to increase/decrease volume using DTMF controls.  In

  opt/freeswitch/conf/autoload_configs/conference.conf.xml

you see the options

  <caller-controls>

    <group name="default">

      <control action="mute" digits="0"/>

      <control action="deaf mute" digits="*"/>

      <control action="energy up" digits="9"/>

      <control action="energy equ" digits="8"/>

      <control action="energy dn" digits="7"/>

      <control action="vol talk up" digits="3"/>

However, there isn't the ability for the user to send these DTMF commands (such as pressing '9' on your phone's keypad).

For now, we would recommend that users who are connecting via Flash audio (not WebRTC audio), make use of the echo test to hear how their audio sounds and use the built-in controls to increase audio gain.

Regards,... Fred

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