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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