VoIP Mailing List Archives

[pauld at versafon.com]

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 using PCMU codec from X-Lite and Linksys 2002.
Is there anything can be tweaked in FS to correct that? Same issue was
with 1.0.2.

_______________________________________________
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 at freeswitch.org]

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 since been fixed in SVN trunk as of yesterday.

/b

On Feb 14, 2009, at 8:37 PM, Paul D. wrote:

[pauld at versafon.com]

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 resampling
involved, and that looks like true since I specifically restricted
codecs in my test SIP equipment.
But the fact is I tried different boxes, same OS centos 5.2 x64, and I
had to bring audio volume and mic level all the way up in X-Lite to
compensate for the difference to * audio,
and in * such volume level sounds like way too high.
FS installed cleanly from scratch, mostly default settings, except some
dialplan/directory additions.

Brian West wrote:

[brian at freeswitch.org]

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 trunk. Let me know how that goes.

/b

On Feb 14, 2009, at 9:02 PM, Paul D. wrote:

[jason at jasonjgw.net]

Brian West <brian@freeswitch.org> wrote:

[brian at freeswitch.org]

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:

[anthony.minessale at g...]

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 sample rates or codecs which you are not because you mentioned it was PCMU.

If you are making a call from x-lite to a linksys using just PCMU there is no transcoding going on at all and it would not be any more or less loud than if the
devices were exchanging media directly because all we would be doing is passing the digital packets across.

I believe you are somehow mistaken in your explanation. There is a good chance that your x-lite has the gain set lower when you are testing FS since that's the only device
in your whole scenario that is capable of adjusting the gain.

If you wish, please get a complete packet capture of a completed call in both situations.


On Sat, Feb 14, 2009 at 8:37 PM, Paul D. <pauld@versafon.com (pauld@versafon.com)> wrote:

[pauld at versafon.com]

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 comparing audio in the call scenarios above
* somehow does noticeably better job, sounds clearer and volume is at
the right level. I am not changing any phone settings of course when
switching between * and FS.
I am not biased towards FS or * at the moment, though FS seems to have a
better designed configuration options and community.
Just wanted to share my experience, and hear some opinions.
Unfortunately I cannot spend whole amount of time investigating this
case now, capturing packets etc., but I will try to do that once I have
time. Meanwhile I will have to stick to * for prod.


Anthony Minessale wrote:

[brian at freeswitch.org]

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:

[brian at freeswitch.org]

Also you didn't try SVN Trunk?

/b

On Feb 15, 2009, at 8:04 PM, Paul D. wrote:

[krice at freeswitch.org]

Paul,

If you are truly having a problem please do get us a full packet trace
including the RTP...

As one of the largest FS users, I can tell you we have not seen this issue
and we interconnect with dozens of different endpoint manufacturers using
FreeSWITCH. (I run tollfreegateway.com an Open SIP to North American
Tollfree TDM termination gateway)

If this problem was wide spread I would suspect that users of several ITSPs
would be complaining and their ITSPs would be be complaining to me.

Now that being said, you're post really smells of a troll.

If it is meant as an honest problem please do get us the trace and we'll be
more than happy to look at it. Also, as was stated earlier if you are
running 1.0.3RC1 then you might see a re-sampling problem in a trans-coding
scenario, this has been resolved and you were advised to run trunk to get
this fix.

As far as your comment on spending too much time to investigate this, all we
have asked for is a simple packet trace... This is something that can be
done in 5 minutes

K

[anthony.minessale at g...]

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.

[jaybinks at gmail.com]

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. <pauld@versafon.com (pauld@versafon.com)> wrote:

[pauld at versafon.com]

Eh??

Ken Rice wrote:

[pauld at versafon.com]

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: