<div>The motivation is to fix the audio distortions.</div><div><br></div><div>The architecture is as follows. One freeswitch acting as a &quot;gateway&quot; with configured sip gateway connected to a VOIP provider. Second freeswitch acting as a &quot;controller&quot; conects to the &quot;gateway&quot;. Incoming call passes the &quot;gateway&quot; to the &quot;controller&quot; where it is handled with ESL. Routing is decided in an ESL application and in its simplest case is forwarded back to the &quot;gateway&quot; through the VOIP provider to another PSTN number. The forwarding is implemented in the &quot;controller&quot; such that uuid_originate is issued and then bridged with the incoming call. No transocding should take place, neither between the&quot;gateway&quot; and VOIP provider nor between the &quot;controller&quot; and the &quot;gateway&quot;.</div>

<div><br></div><div>Although no transcoding takes place and the system is idle, still there are little distortions in the audio. Distortions that don&#39;t exist if I connect to the VOIP directly with a SIP phone.</div><div>

<br></div><div>I&#39;m looking for a way to find out, where the distortion is introduced.</div><br clear="all">Hynek<br><br>
<br><br><div class="gmail_quote">On Wed, Nov 2, 2011 at 1:40 PM, Sammy Govind <span dir="ltr">&lt;<a href="mailto:govoiper@gmail.com">govoiper@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

Please explain your query bit more, keeping in mind the ideal world scenario as your&#39;s, what exactly you require to know.<div><div></div><div class="h5"><br><br><div class="gmail_quote">On Wed, Nov 2, 2011 at 5:09 PM, Hynek Cihlar <span dir="ltr">&lt;<a href="mailto:hynek.cihlar@gmail.com" target="_blank">hynek.cihlar@gmail.com</a>&gt;</span> wrote:<br>


<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div>In this case, it&#39;s the &quot;ideal&quot; world. Testing environment, all media servers on one OS, one native server box, CPU idle, no load, no transcoding. One media switch is connected to a VOIP provider though through SIP. But the provider sits in the same telehouse and again no transcoding takes place.</div>




<div><br></div><div>Any idea where to direct the investigation?</div><br clear="all"><font color="#888888">Hynek</font><div><div></div><div><br><br>
<br><br><div class="gmail_quote">On Wed, Nov 2, 2011 at 6:42 AM, Sammy Govind <span dir="ltr">&lt;<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">





Network latency, transcoding processing, jitter handling, call session establishment on each Media-Server/switches etc all add up in overall call quality factor.<br><br><div class="gmail_quote"><div><div></div><div>
On Tue, Nov 1, 2011 at 11:39 PM, Hynek Cihlar <span dir="ltr">&lt;<a href="mailto:hynek.cihlar@gmail.com" target="_blank">hynek.cihlar@gmail.com</a>&gt;</span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div></div><div>Would anybody know, if the call quality could be dependant on the<br>
number of media switches the call is routed through?<br>
<br>
Currently I&#39;m testing a system with several freeswitches in the call<br>
way (up to 4) and I&#39;m noticing very short gaps, not very disturbing<br>
but noticable.<br>
<br>
What&#39;s your experience?<br>
<br>
Sent from my mobile device<br>
<br>
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