<span style="font-size: small;">Hi Jay</span><br /><br /><span style="font-size: small;">I use the following dialstring to add a jitter buffer :</span><br /><span style="font-size: small;"> <action application="bridge" data="{jitterbuffer_msec=80}sofia/gateway/siptrunk/$1@1.2.3.4"/></span><br /><br /><span style="font-size: small;">The wiki says the PLC is enabled by default when using a jitter buffer, but you can try this to be extra sure:</span><br />
<pre><span style="font-size: small;"><action application="set" data="sip_jitter_buffer_plc=true"/></span><br /><br /><span style="font-size: small;">Please let us know how you resolve this, as its a very interesting topic to those of us who live with not ideal connectivity.</span><br /><br /><br /><span style="font-size: small;">Regards</span><br /><span style="font-size: small;">Clive</span><br /><br /><br /></pre>
<br /><br />On Wed, 29 Jan 2014 09:00:12 +1000 jay binks <jaybinks@gmail.com> wrote<br /><br />
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<div dir="ltr">So I have a freeswitch usecase that Im having trouble solving and would like the collective help of the community.
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<div>I have a customer who has a commercial PBX on site ... ( Supports SIP )</div>
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<div>However their network connectivity is quite bad at times, and there are no other options for connectivity. ( lets say they are a mine site or something remote like that )</div>
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<div>SIP calls on this link sound pretty bad when they experience network issues ( mainly jitter & some loss ), BUT Skype calls sound pretty good. ( not perfect, but a LOT better than say G711, G729 or any other open standard VOIP Call )</div>
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<div>What we considered doing is deploying a freeswitch box on their site, and one in a DC with reliable connectivity. have their PBX send G711 calls to the onsite freeswitch, and have freeswitch transcode this call to OPUS / SILK. send the OPUS / SILK over the bad network to the co-located freeswitch box, who will then trancode back to G711 and set the call to the PSTN network like normal.</div>
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<div>The intention here is to leverage the Packet Loss Concealment baked into OPUS / SILK</div>
<div>as I imagine that is what gives skype its ability to hide the audible effects of loss.</div>
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<div>So far I have only set this up in a lab and used WANem to simulate loss, jitter, out of order packets. but ive found OPUS / SILK to be no better than G711 at covering loss.</div>
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<div>I even tried with a 2 sec jitter buffer on both FS instances ( on each end ), which made no difference.</div>
<div>I was expecting the jitter buffer to add more delay to the call, but it didnt even do this ... maybe I misconfigured something in the jitter buffers.</div>
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<div>anyways.. I would like some help with this, has anyone tried doing anything similar to this and had success ?</div>
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<div>--</div>
<div>Sincerely<br /><br />Jay Binks</div>
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