Which is 40ms... which is what FS tells you. FS will look at the amount *actually* received, instead of the amount the client claimed it will sent. It auto-adjusts to that amount, and displays the warning message because your client is either misconfigured or broken (some SIP phone implementations lie).<br>
<br>Because it auto-adjusts it shouldn&#39;t be an issue, but it could also affect quality so a warning is given just in case so that you can fix the problem on the client.<br><br>-Steve<br><br><br><div class="gmail_quote">
On 26 August 2010 04:30, Michael Jerris <span dir="ltr">&lt;<a href="mailto:mike@jerris.com">mike@jerris.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div style="word-wrap: break-word;">20 ms of ulaw would be 160 bytes, not 320.<div><br></div><div>Mike</div><div><div></div><div class="h5"><div><br><div><div>On Aug 25, 2010, at 4:46 PM, Johny Kadarisman Kwan wrote:</div>
<br><blockquote type="cite">I&#39;m trying to convert a proprietary audio stream into sip/rtp compatible. At this point, i&#39;m able to pass sip negotiation with FS, and trying to stream PCMU/8000 codec. But something not right on my end, and FS complained about the ptime settings.<div>

I&#39;m sending 320 bytes ulaw payload on every rtp packet. Is that the right amount? appreciate if anyone could point some info on these topics.</div><div><br></div><div>Thanks,</div><div>JK<br><div><br></div><div>===============</div>

<div><div><b>2010-08-25 16:36:58.142328 [WARNING] mod_sofia.c:1013 We were told to use ptime 20 but what they meant to say was 40</b></div><div><b>This issue has so far been identified to happen on the following broken platforms/devices:</b></div>

<div><b>Linksys/Sipura aka Cisco</b></div><div><b>ShoreTel</b></div><div><b>Sonus/L3</b></div><div><b>We will try to fix it but some of the devices on this list are so broken,</b></div><div><b>who knows what will happen..</b></div>

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