Thanks Anthony!

I tried adding this var and also tried uncommenting the suppress_cng line in my sip profile.  It did not seem to make a difference.  I pointed my client code to your sample and it worked fine - no noise, so it's not a problem with the client side.

Do I need to upgrade to the latest master?  I am using master from 5/22 version 1.5.13b

Would it help if gave you my test ws server and sip address so you could reproduce this?  

Is it possible that the problem is with my AWS instance or CentOS 5.9?




On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
I set up your test ext here:
https://webrtc.freeswitch.org/sipjs/index.html#8765

And it seems to work.

Try this in vars.xml

<X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> 

I don't think the silence suppression works very well in the browser.





On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
In case it was not clear from the previous email - this only happens with conferences.  If after the beep, instead of placing the call into a conference, I just play some music, there are no problems.


On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi guys,

I am connecting to FreeSWITCH using WebRTC and after a user calls in, I play a beep and put them into a conference.  Two strange things happen after that:

1. There is a noise in the user's leg that someone described like a wind in the tunnel.  The noise lasts for several seconds, then the user can hear the hold music.  The noise does not seem to happen if I don't play the beep before the conference.

2. About 3 seconds into the call, there is another very short beep - like a small portion of the original beep that the user hears.

This absolutely does not happen with softphones - only with WebRTC.  I tried it with Chrome and Opera - same thing is happening on both.

I tried increasing the answer delay from 2 seconds to 3 and the noise now happens more rarely but the extra beep is still there almost every time.  I am using JsSIP 3.0 but since it only handles signalling I don't think the problem is with the JS library.

Has anyone run into this?  Any advice?

Here is the relevant excerpt from my dialplan.

<condition field="destination_number" expression="^conf-(\S+)$">
        <action application="set" data="answer_delay=3000"/>
      <action application="answer"/>
        <action application="playback" data="tone_stream://%(1000,0,600)" />
        <action application="conference" data="$1@default"/>
</condition>



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--
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

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

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

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