<div>I have two different network setups, and have seen similar lag on both.</div>
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<div>The first is my home testbed. I'm connected to the internet through a home router and then a cablemodem. The home environment is pretty spare, of course. 2 machines and a couple of T-mobile cell phones with their SIP communication is all that goes through there. I have used my cell phone, a couple of different softphones, Gizmo call-ins, and regular PSTN calls. The worst lag is the T-mobile cell phones, but I'm happy to write that off as T-mobile's problem if we'd like.</div>
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<div>The second is the debug server environment on the systems where the conference product will eventually reside. The system is very complex, as it is already running a major hosted PBX service written years ago. I'm afraid all of the details of this system are beyond me, but I know that it includes a PSTN gateway, more T1s than I can count, and I'm having to split the RTP and SIP packets on separate ports for security and organizational purposes. For call-ins, I have used T-mobile again and regular PSTN, no softphones (yet). Obviously, this is the important environment, and the PSTN lag is somewhere around 500-700 ms (subjective).</div>
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<div>So am I correct in understanding that this is not a common issue, then, and that something can theoretically be done to help it?<br><br></div>
<div class="gmail_quote">On Tue, Jun 16, 2009 at 11:35 AM, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Can you describe your networking environment a bit? One thing that can affect the latency of your voice traffic is your network infrastructure. If you can isolate FS and some phones on a separate, controlled network then possibly you can start narrowing it down to other factors.<br>
<br>-MC<br><br>
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<div class="h5">On Tue, Jun 16, 2009 at 10:51 AM, Bradley Brashier <span dir="ltr"><<a href="mailto:bjbrashier@gmail.com" target="_blank">bjbrashier@gmail.com</a>></span> wrote:<br></div></div>
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<div class="h5">I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy).<br>
<br>I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call.<br>
<br>My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace? <br>
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