-users] FS SIP audio quality?
I re-tested calls to VM replacing some of FS prompts with * ones, and it
appears that * sounds were recorded with a better quality/higher volume,
so FS itself has nothing to do with that. That's solved. :-)
I am going to double check all the equipment we used for tests
jay binks jaybi...@gmail.com wrote:
Back in November, Brian ( BKW ) was raising money to get new sounds recorded
...
intending to have them for the 1.0.2 release..
I wonder if they made it in, or if they are still coming ...
Release 1.0.7 of the sound files was made available soon
: Re: [Freeswitch-users] FS SIP audio quality?
I re-tested calls to VM replacing some of FS prompts with * ones, and it
appears that * sounds were recorded with a better quality/higher volume,
so FS itself has nothing to do with that. That's solved. :-)
I am going to double check all
Subject: Re: [Freeswitch-users] FS SIP audio quality?
I re-tested calls to VM replacing some of FS prompts with * ones, and it
appears that * sounds were recorded with a better quality/higher volume,
so FS itself has nothing to do with that. That's solved. :-)
I am going to double
Paul D. wrote:
I re-tested calls to VM replacing some of FS prompts with * ones, and it
appears that * sounds were recorded with a better quality/higher volume,
so FS itself has nothing to do with that. That's solved. :-)
There's a long history of people in A/B listening tests reporting
David Knell napsal(a):
There's a long history of people in A/B listening tests reporting louder
as sounding
better on the same source material - even if the additional volume isn't
detectable
as such.
Yes, you are right :-) And therefor a lot of (nearly all of) European
TelCo operator
Maybe the sox script brian uses to downsample the files has a problem.
What if you download the 48k package (original) and listen to that?
On Tue, Feb 17, 2009 at 6:35 AM, David Knell d...@3c.co.uk wrote:
Paul D. wrote:
I re-tested calls to VM replacing some of FS prompts with * ones, and
Eh??
Ken Rice wrote:
Paul,
Now that being said, you're post really smells of a troll.
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I was trying to send tcp dumps today, but the message was rejected
because of its size (zipped). How do I send them?
Anthony Minessale wrote:
The typing it takes to start a pcap of each call and email them is
less than you have typed thusfar.
Please just take the captures and send them to
You can send them directly to me br...@freeswitch.org
Thanks,
/b
On Feb 16, 2009, at 6:08 PM, Paul D. wrote:
I was trying to send tcp dumps today, but the message was rejected
because of its size (zipped). How do I send them?
Anthony Minessale wrote:
The typing it takes to start a pcap
On Mon, Feb 16, 2009 at 4:08 PM, Paul D. pa...@versafon.com wrote:
I was trying to send tcp dumps today, but the message was rejected
because of its size (zipped). How do I send them?
Can you put them on a server where the devs can use wget or a browser
to download them?
-MC
: [Freeswitch-users] FS SIP audio quality?
Eh??
Ken Rice wrote:
Paul,
Now that being said, you're post really smells of a troll.
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I re-tested calls to VM replacing some of FS prompts with * ones, and it
appears that * sounds were recorded with a better quality/higher volume,
so FS itself has nothing to do with that. That's solved. :-)
I am going to double check all the equipment we used for tests, like
headphones,
it's digital audio. The only thing doing sampling and reconstruction of the
signal are the phones. The audio files have been captured long ago from the
microphone in the studio.
We do nothing to alter the volume of the audio signal or manipulate it in
any way unless you are transcoding between
Well, I tried several call scenarios:
1. Call from X-Lite or Linksys to VM.
2. Call from X-Lite or Linksys to a conference.
3. Call from X-Lite or Linksys to a PSTN number via Gafachi and CallWithUs.
I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise
grade Intel server. So just
I'm not able to reproduce this issue.. can you verify the codecs are
what you think they are on both Asterisk and FreeSWITCH.
/b
On Feb 15, 2009, at 8:04 PM, Paul D. wrote:
Well, I tried several call scenarios:
1. Call from X-Lite or Linksys to VM.
2. Call from X-Lite or Linksys to a
Also you didn't try SVN Trunk?
/b
On Feb 15, 2009, at 8:04 PM, Paul D. wrote:
I have now * 1.6.5 and FS 1.0.3RC1
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Subject: Re: [Freeswitch-users] FS SIP audio quality?
Well, I tried several call scenarios:
1. Call from X-Lite or Linksys to VM.
2. Call from X-Lite or Linksys to a conference.
3. Call from X-Lite or Linksys to a PSTN number via Gafachi and CallWithUs.
I have now * 1.6.5 and FS
The typing it takes to start a pcap of each call and email them is less than
you have typed thusfar.
Please just take the captures and send them to us to examine. That's all. If
you have a real issue we would like to address it.
On Feb 15, 2009 8:06 PM, Paul D. pa...@versafon.com wrote:
Well, I
another thing to try here...
is to put FS in RTP proxy and bypass mode.
http://wiki.freeswitch.org/wiki/Bypass_Media
it would be interesting to see if your still experiencing this problem in
either of those 2 modes.
Jay
On Mon, Feb 16, 2009 at 12:04 PM, Paul D. pa...@versafon.com wrote:
Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Sip-to-Sip call, or
call to VM prompt, or call via gateway to PSTN - FS audio volume level
(should I say gain?) seems noticeably lower than on *, this may be a
reason that FS audio seems to be subpar, more noise less clear. Test
calls made
I haven't ever experienced this issue can you maybe elaborate on the
issue a little more? We usually hear that the audio quality is much
better... have you tried latest SVN trunk? If resampling was involved
it might cause some audio issues but those were usually gain issue and
that has
I am not sure what else I can add to that, I would love to elaborate
more if you ask anything specific.
I haven't tried the latest trunk, but since there's no difference
between 1.0.2 and 1.0.3RC1 in audio quality I don't think
it make sense trying. From what I see in FS logs there's no
What is odd some people have reported the same issue with Asterisk. I
would like to get to the bottom of it but nobody can provide any more
detail on what might be going on and I haven't experienced this issue
with the 30 or so phones I have on my desk I highly recommend you
try SVN
Brian West br...@freeswitch.org wrote:
What is odd some people have reported the same issue with Asterisk. I
would like to get to the bottom of it but nobody can provide any more
detail on what might be going on and I haven't experienced this issue
with the 30 or so phones I have on my
This was a problem with the resampler which was replaced... we use the
resampler in Speex now which will not exhibit the problem.
/b
On Feb 14, 2009, at 9:18 PM, Jason White wrote:
I sometimes get audio distortion in the above situation if anyone
speaks too
loudly. I suspect clipping
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