<div dir="ltr">In practice, internally there is little reason to have each machine actually handling the media. Set bypass_media=true before bridging the call.<div>This doesn't address the issue, though, if there is one.</div>
<div><br></div><div>Can you see which machine introduces the distortion? e.g. Is it ok if you only use 3?<br><div><br clear="all"><div dir="ltr"><span style="font-family:Verdana, Arial, Helvetica, sans-serif"><span style="font-size:small">-Avi</span></span></div>
<br><div class="gmail_quote">On Wed, Nov 2, 2011 at 3:15 PM, Sammy Govind <span dir="ltr"><<a href="mailto:govoiper@gmail.com">govoiper@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Please be sure that no-transcoding is done. Make sure the codecs of inbound and outbound legs are same on provider<->gateway and then gateway<->controller and way back. <div>If transcoding isn't the case, in my experiences virtual machines show these little distortions in call.<div>
<div class="h5"><br>
<br><div class="gmail_quote">On Wed, Nov 2, 2011 at 5:56 PM, Hynek Cihlar <span dir="ltr"><<a href="mailto:hynek.cihlar@gmail.com" target="_blank">hynek.cihlar@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>The motivation is to fix the audio distortions.</div><div><br></div><div>The architecture is as follows. One freeswitch acting as a "gateway" with configured sip gateway connected to a VOIP provider. Second freeswitch acting as a "controller" conects to the "gateway". Incoming call passes the "gateway" to the "controller" where it is handled with ESL. Routing is decided in an ESL application and in its simplest case is forwarded back to the "gateway" through the VOIP provider to another PSTN number. The forwarding is implemented in the "controller" such that uuid_originate is issued and then bridged with the incoming call. No transocding should take place, neither between the"gateway" and VOIP provider nor between the "controller" and the "gateway".</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't exist if I connect to the VOIP directly with a SIP phone.</div><div>
<br></div><div>I'm looking for a way to find out, where the distortion is introduced.</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 1:40 PM, Sammy Govind <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></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's, what exactly you require to know.<div><div></div><div><br><br><div class="gmail_quote">On Wed, Nov 2, 2011 at 5:09 PM, Hynek Cihlar <span dir="ltr"><<a href="mailto:hynek.cihlar@gmail.com" target="_blank">hynek.cihlar@gmail.com</a>></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's the "ideal" 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"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></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"><<a href="mailto:hynek.cihlar@gmail.com" target="_blank">hynek.cihlar@gmail.com</a>></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'm testing a system with several freeswitches in the call<br>
way (up to 4) and I'm noticing very short gaps, not very disturbing<br>
but noticable.<br>
<br>
What's your experience?<br>
<br>
Sent from my mobile device<br>
<br>
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