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.
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:
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.
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:
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:
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.
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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:
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 <brian@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 desk .... I highly recommend you
A data point that may or may not be helpful: if I set up PortAudio on FreeSWITCH and call an Asterisk conference from there, the audio is significantly louder than a comparable SIP call with another FreeSWITCH box at the other end.
try SVN trunk. Let me know how that goes. ;)
I'll recreate the above scenario with SVN trunk (I've just built rev. 12018), and report if there is still a problem. I sometimes get audio distortion in the above situation if anyone speaks too loudly. I suspect clipping somewhere in the audio processing.
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 somewhere in the audio processing.
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> 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 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.
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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:
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 <mailto:pauld@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 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.
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 ------------------------------------------------------------------------
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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 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.
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:
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 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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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:
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 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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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:
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:
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 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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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>wrote:
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:
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:
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 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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-- Sincerely Jay
jay binks <jaybinks@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 thereafter, which I understand includes the new material that was recorded. I might be wrong, though.
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 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
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 (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
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> wrote:
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 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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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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
From: "Paul D." <pauld@versafon.com> Reply-To: <freeswitch-users@lists.freeswitch.org> Date: Sun, 15 Feb 2009 21:04:14 -0500 To: <freeswitch-users@lists.freeswitch.org> 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 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:
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 <mailto:pauld@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 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.
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 ------------------------------------------------------------------------
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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
From: "Paul D." <pauld@versafon.com> Reply-To: <freeswitch-users@lists.freeswitch.org> Date: Mon, 16 Feb 2009 19:06:18 -0500 To: <freeswitch-users@lists.freeswitch.org> Subject: Re: [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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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> 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 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:
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 <mailto:pauld@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 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 <mailto: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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> <mailto:MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com>
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IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> <mailto:sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org>
iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org>
pstn:213-799-1400 ------------------------------------------------------------------------
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-- Sincerely Jay
participants (11)
-
Anthony Minessale
-
Brian West
-
David Knell
-
Giovanni Maruzzelli
-
Jason White
-
jay binks
-
Ken Rice
-
Ken Rice
-
kokoska rokoska
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Nik Middleton
-
Paul D.