well waste bandwidth became obsolete on oct 4th<div><br></div><div>You could test your theory by using a 30ms/20ms phone instead.</div><div>You should really remove variables until you come up with something.</div><div><br>
</div><div>Basically the default FS conf on <a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a> is running the default params with 200 as default energy.</div><div>So call that first and see if you get the same issue because we have 30+ people call there every wednesday without incident and we literally spend like 12 hours a day on the same server for private development.</div>
<div><br></div><div>Then setup a vanilla out of the box FS build and try exten 3000</div><div>The conference. and see if you get your problem, try other phones etc.</div><div><br></div><div>look at all the variables to see what you may be doing differently than most people who do not experience your issue.</div>
<div><br></div><div><br></div><div> </div><div><br></div><div><br><br><div class="gmail_quote">2011/11/3 Roland Hänel <span dir="ltr"><<a href="mailto:roland@haenel.me">roland@haenel.me</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div>Yes, I am on a current GIT (about 2 weeks old), and I have waste-bandwidth for every subscriber, and a packet capture confirms that FS sends RTP permanently. So jitter buffers shouldn't be an issue here. </div><div>
<br></div><div>Even *if* jitter buffers would be recharging, that would only lead to a small delay, but not to audio being clipped (i.e., lost), right?</div><div><br></div><div>My packet traces show that there are different ptime's involved (FreeSwitch is on 20ms by default, some phones have 30ms). Could that contribute to the issue?</div>
<div><div></div><div class="h5">
<div><br></div><div>Roland</div><br><br><div class="gmail_quote">2011/11/3 Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
it's not,<div><br></div><div>most likely you have a jitterbuffer on your phone that is recharging when you start talking.</div><div>are you on the latest GIT where we have the waste bandwidth flag on permanently for all callers?</div>
<div>To avoid problems like this conferences now always send rtp to people even when they are not getting any useful input.</div><div><div></div><div><div> <br><br><div class="gmail_quote"><a href="tel:2011%2F11%2F3" value="+492011113" target="_blank">2011/11/3</a> Roland Hänel <span dir="ltr"><<a href="mailto:roland@haenel.me" target="_blank">roland@haenel.me</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Anthony,<div><br></div><div>The problem is not that audio is clipped because its "too silent", so adjusting the value seems to be the wrong handle for this problem.</div>
<div><br></div><div>It seems that the time needed to make the decision (regardless of the energy threshold value) is the core of the issue, i.e. the level may bei 100, 200 or 300, it takes some XX ms to detect talking and all audio that happens before the detection gets lost. Right?</div>
<div><br></div><div>Greetings,</div><div>Roland</div><div><br></div><div><div><div></div><div><br><br><div class="gmail_quote"><a href="tel:2011%2F11%2F3" value="+492011113" target="_blank">2011/11/3</a> Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>The setting is an arbitrary number of the energy score required to count as talking.</div></div><div><br></div>
<div><br></div>check your environments, its very specific to the surroundings.<div>We run it at 200 in the main FS conf and its rarely an issue.</div><div><div></div><div>
<div><br></div><div><br></div><div><br></div><div><br><div class="gmail_quote"><a href="tel:2011%2F11%2F3" value="+492011113" target="_blank">2011/11/3</a> Roland Hänel <span dir="ltr"><<a href="mailto:roland@haenel.me" target="_blank">roland@haenel.me</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<p>Setting it to 10 or so doesn't help, it seems that just the threshold of detection is changed (ok, that's what is expected), not the time until the decision is made.</p>
<p>The audio clipping is a timing issue. <br>
Does there exist something like a "minimum time" that this algorithm needs to detect talking after silence? What happens to the audio that was sent by that member during this time, does.the algorithm discard that?</p>
<p>Greetings,<br>
Roland</p>
<div class="gmail_quote">Am 03.11.2011 18:20 schrieb "Anthony Minessale" <<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>>:<div><div></div><div><br type="attribution">
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
yes? set it to something between what it was and 0 ?<div><br></div><div> <param name="energy-level" value="300"/></div><div><br></div><div>try <a href="tel:200%20100%2050" value="+4920010050" target="_blank">200 100 50</a> ?<br>
<br><div class="gmail_quote"><a href="tel:2011%2F11%2F3" value="+492011113" target="_blank">2011/11/3</a> Roland Hänel <span dir="ltr"><<a href="mailto:roland@haenel.me" target="_blank">roland@haenel.me</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<div><br></div><div>During thorough testing of our conference solution which is based on FreeSwitch / mod_conference, we noticed some 'audio clipping' during the conference. Typically, if a conference member didn't have audio for some time and then begins to speak, the first syllable will be clipped and cannot be heard by the other conference participants.</div>
<div><br></div><div>We ruled out a number of possible causes, and finally discovered that the energy level detection algorith seems to be the source of the problem. Setting the energy level for all participants to "0" heals the problem (of course now the noise might be higher since every participant's noise contributes to the total noise), but this setting is not very useful since the "talking detection" seems to rely on the same level, i.e. with energy "0", all members are marked talking all the time (which effectively renders the "talking" display useless).</div>
<div><br></div><div>Is there any chance to tune the energy level detection time (!), or to de-couple talking detection an energy level detection in FreeSwitch?</div><div><br></div><div>Best regards,</div><div>Roland</div>
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