<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Going back a step, to where Jon was seeing more packets than there<div>should have been, I've just encountered a similar issue having upgraded</div><div>to the latest, from what was probably a fairly old release - months old,</div><div>rather than weeks.</div><div><br></div><div>I've got two FS boxes (let's call them FS1 and FS2), each of which are</div><div>plumbed in to carrier C. There's an IVR service running on FS1; FS2</div><div>bridges any calls which it gets for said IVR over to FS1. What I've just had</div><div>is:</div><div>- calls from C to FS1 directly work fine;</div><div>- calls from C to FS2, thence to FS1 were silent. Looking at a capture from</div><div>FS2, everything looks OK except the RTP between FS1 and FS2. On answer,</div><div>there's a prompt played. What I see is three packets in a lump from FS1, then</div><div>four packets sent back from FS2 to FS1, four packets in a lump from FS1, then</div><div>five going back from FS2 to FS1, and so on.</div><div><br></div><div>The lumps are 20ms apart (codec is G711 with 20ms packets) - what seems to be</div><div>happening is that FS2 sends FS1 back the packets received from it unchanged</div><div>plus an extra packet which has arrived from C in the meantime.</div><div><br></div><div>FS2 ought to be sending these packets to C instead; it sends C nothing.</div><div><br></div><div>I've made the problem go away by commenting out the bit in switch_rtp.c which</div><div>auto-adjusts addresses (around line 1280.)</div><div><br></div><div>All of the machines have public IPs; there's not a NAT in sight.</div><div><br></div><div>I'll have a further look in the morning.</div><div><br></div><div>--Dave</div><div><br></div><div><div><br class="Apple-interchange-newline"><blockquote type="cite"><div dir="ltr">%(60000,0,300) means to generate a 60 second long 300hz tone <br>%(5,0,300) means a 5 ms long 300hz tone<br><br>if you are just trying to send a tone you are better off with<br><action application="gentones" data="%(1000,0,300)|60"/><br> <br>which only generates 1 second of audio then buffers and loops it via the application<br>rather than allocating enough room for 60 seconds of signed linear audio and generating<br>the whole 60 seconds into memory for no reason vs 1 second sample looped 60 times.<br> <br>No matter what you do it will not effect the bandwidth used, it's a factor of what codec you are using.<br><br><br><div class="gmail_quote">On Mon, Oct 6, 2008 at 4:15 AM, Jon Bruel <span dir="ltr"><<a href="mailto:jbr@consiglia.dk">jbr@consiglia.dk</a>></span> wrote:<br> <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Resolved: I have made further tests, and my final conclusion is that the<br> previous stated test results were screwed by the application 'gentones'.<br> This application does in some cases send more rtp than expected. If I<br> used:<br> <action application="gentones" data="%(5,0,300)"/><br> <action application="gentones" data="%(5,0,300)"/><br> <action application="gentones" data="%(60000,0,300)"/><br> the expected rtp of 8600 kB/s was transmitted. If I used<br> <action application="gentones" data="%(60000,0,300)"/><br> <action application="gentones" data="%(5,0,300)"/><br> <action application="gentones" data="%(5,0,300)"/>.<br> the rtp was 34600 kB/s, and the memory is heavily consumed. The only<br> difference being the sequence of the gentones commands. I don't know if<br> this is the expected behaviour of 'gentones' or not, but it certainly<br> screwed up the results previously posted. /Jon<br> <div><div></div><div class="Wj3C7c"><br> <br> _______________________________________________<br> Freeswitch-users mailing list<br> <a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br> </div></div></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br> <br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br> IRC: <a href="http://irc.freenode.net">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br><a href="http://iax:guest@conference.freeswitch.org/888">iax:guest@conference.freeswitch.org/888</a><br> <a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>pstn:213-799-1400<br> </div> _______________________________________________<br>Freeswitch-users mailing list<br><a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org<br></blockquote></div><br></div></body></html>