<div dir="ltr">are you doing anything with bypass_media?<br>We had another guy with something similar and we have not been able to nail it down besides<br>he was doing a media call that shifted to bypass_media mode.<br><br>
If you have a situation where it happens every time a pcap and or FS console on debug of it happening would be nice.<br>It seems odd that the auto adjust code is being triggered considering the design says that if the leg from FS1 -> C gets a packet<br>
from some other ip than that of C that it adjusts to that. So it would appear that somehow one of the packets from FS2 to FS1<br>ends up being picked up by the rtp socket that was meant for the FS1->C leg. (maybe a race in the port allocator giving both legs the same rtp port or something funny?) <br>
<br>say your phone is A<br><br>A -> FS1 (recv port 1000)<br>FS1->FS2 (recv port 1002)<br>FS2->C (recv port 1004)<br><br>either port 1004 is somehow getting the packet meant for 1002 which would make no sense<br>or somewhere in here both boxes picked the same port or the packet is somehow multicasting to both ports which would make no sense either.<br>
<br>I would like to figure it out though if you could provide the details including the dp and all the attrs you set etc.<br><br><br>BTW, you can turn off RTP auto ADJ both with a channel varable and a profile param (i forgot the name but it's on the wiki)<br>
<br><br><br><br><br><br><br><br><br><div class="gmail_quote">On Mon, Oct 6, 2008 at 9:25 PM, David Knell <span dir="ltr"><<a href="mailto:dave@3c.co.uk">dave@3c.co.uk</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;">
<div style="">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><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" target="_blank">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><br> <br> _______________________________________________<br> Freeswitch-users mailing list<br>
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