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msc at freeswitch.org Guest
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Posted: Mon Feb 16, 2009 7:18 pm Post subject: [Freeswitch-users] FS SIP audio quality? |
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On Mon, Feb 16, 2009 at 4:08 PM, Paul D. <pauld@versafon.com> wrote:
Quote: | 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
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brian at freeswitch.org Guest
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krice at suspicious.org Guest
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Posted: Mon Feb 16, 2009 9:22 pm Post subject: [Freeswitch-users] FS SIP audio quality? |
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If it wasn't meant as a troll I personally, publically, and directly
apologize to you...
What I see all the time is people want everything for free and then think
its the developers responsibility to give away free tech support on this
software which is free in the first place.
Tony and his crew work on FreeSWITCH as much to feed their families as they
do to have an open platform that anyone can use
K
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pauld at versafon.com Guest
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Posted: Mon Feb 16, 2009 10:36 pm Post subject: [Freeswitch-users] FS SIP audio quality? |
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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, telephone sets, cables since I am almost convinced that
there's nothing in FS which can produce effects I observe.
I will post back if I find anything wrong, appreciate everybody's help
with this.
Brian West wrote:
Quote: | 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:
Quote: | 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.
|
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nik.middleton at noble... Guest
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Posted: Tue Feb 17, 2009 5:08 am Post subject: [Freeswitch-users] FS SIP audio quality? |
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For what it's worth, using Asterisk recordings, I found FS to be better
than when played on an Asterisk system.
I came to the same conclusion early on that the included prompts with FS
were of a relatively poor nature. Not volunteering to record new ones,
but they do let the product down, as they lead to discussions as below.
Regards,
-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Paul
D.
Sent: 17 February 2009 03:33
To: freeswitch-users@lists.freeswitch.org
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 check all the equipment we used for tests, like
headphones, telephone sets, cables since I am almost convinced that
there's nothing in FS which can produce effects I observe.
I will post back if I find anything wrong, appreciate everybody's help
with this.
Brian West wrote:
Quote: | 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:
Quote: | 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.
|
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| UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Quote: | http://www.freeswitch.org
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gmaruzz at celliax.org Guest
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Posted: Tue Feb 17, 2009 5:21 am Post subject: [Freeswitch-users] FS SIP audio quality? |
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There is also another side to make mimd to: the Asterisk sounds you
hear more often (the demo ones) are very long ones.
The ones of the FS demo are very very short (many times just one word)
and concatenated with the insertion of sleeps.
That is probably someway altering the equation between user experiences
Sincerely,
Giovanni Maruzzelli
=========================================
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039
On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton
<nik.middleton@noblesolutions.co.uk> wrote:
Quote: | For what it's worth, using Asterisk recordings, I found FS to be better
than when played on an Asterisk system.
I came to the same conclusion early on that the included prompts with FS
were of a relatively poor nature. Not volunteering to record new ones,
but they do let the product down, as they lead to discussions as below.
Regards,
-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Paul
D.
Sent: 17 February 2009 03:33
To: freeswitch-users@lists.freeswitch.org
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 check all the equipment we used for tests, like
headphones, telephone sets, cables since I am almost convinced that
there's nothing in FS which can produce effects I observe.
I will post back if I find anything wrong, appreciate everybody's help
with this.
Brian West wrote:
Quote: | 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:
Quote: | 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.
|
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
| UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Quote: | http://www.freeswitch.org
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jaybinks at gmail.com Guest
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Posted: Tue Feb 17, 2009 5:29 am Post subject: [Freeswitch-users] FS SIP audio quality? |
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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 ...
Jay
On Tue, Feb 17, 2009 at 8:19 PM, Giovanni Maruzzelli <gmaruzz@celliax.org (gmaruzz@celliax.org)> wrote:
Quote: | There is also another side to make mimd to: the Asterisk sounds you
hear more often (the demo ones) are very long ones.
The ones of the FS demo are very very short (many times just one word)
and concatenated with the insertion of sleeps.
That is probably someway altering the equation between user experiences
Sincerely,
Giovanni Maruzzelli
=========================================
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039
On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton
<nik.middleton@noblesolutions.co.uk (nik.middleton@noblesolutions.co.uk)> wrote:
Quote: | For what it's worth, using Asterisk recordings, I found FS to be better
than when played on an Asterisk system.
I came to the same conclusion early on that the included prompts with FS
were of a relatively poor nature. Not volunteering to record new ones,
but they do let the product down, as they lead to discussions as below.
Regards,
-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)
[mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] On Behalf Of Paul
D.
Sent: 17 February 2009 03:33
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
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 check all the equipment we used for tests, like
headphones, telephone sets, cables since I am almost convinced that
there's nothing in FS which can produce effects I observe.
I will post back if I find anything wrong, appreciate everybody's help
with this.
Brian West wrote:
Quote: | 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:
Quote: | 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.
|
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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--
Sincerely
Jay |
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jason at jasonjgw.net Guest
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Posted: Tue Feb 17, 2009 5:38 am Post subject: [Freeswitch-users] FS SIP audio quality? |
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jay binks <jaybinks@gmail.com> wrote:
Quote: | 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 thereafter, which I
understand includes the new material that was recorded. I might be wrong,
though.
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dave at 3c.co.uk Guest
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Posted: Tue Feb 17, 2009 7:39 am Post subject: [Freeswitch-users] FS SIP audio quality? |
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Paul D. wrote:
Quote: | 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 louder
as sounding
better on the same source material - even if the additional volume isn't
detectable
as such.
Which, I guess, explains my 25 years of going to Motorhead gigs.
--Dave
--
David Knell, Director, 3C Limited
T: 020 8114 5002 F: 020 3002 7257 M: 07773 800623
http://www.3c.co.uk
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kokoska.rokoska at pos... Guest
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Posted: Tue Feb 17, 2009 8:47 am Post subject: [Freeswitch-users] FS SIP audio quality? |
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David Knell napsal(a):
Quote: | 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 (TDM, not VoIP) normalize their messages to -3 dB instead
of -6 dB.
And one of them (yes, you guess it right, it is Telefonica O2
normalize recordings to 0 dB. And it is VERY loud
Best regards,
kokoska.rokoska
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anthony.minessale at g... Guest
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Posted: Tue Feb 17, 2009 8:56 am Post subject: [Freeswitch-users] FS SIP audio quality? |
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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 <dave@3c.co.uk (dave@3c.co.uk)> wrote:
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400 |
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