<div>If you have access to the clients machine, you can install a network monitoring tool there (Wireshark will be fine) and check OUTGOING RTP stream during the call.</div><div><br></div><div>Make sure that RTP is sent to the port described in FreeSWITCH's OK reply (for example 17252, from pcap file you provided, line #10).</div>
<div><br></div><div>Also check on the FS server, if it receives packets to this stream - use "ngrep port xxxxx" command.</div><div><br></div><div>If data arrives, then problem is in FreeSWITCH (almost unlikely). If not, then you will check further at the routers in between.</div>
<div><br></div><div>Maybe some router is doing some manipulations with the port number, so data is lost before arriving to FS.</div><div><br></div><div>Vitalie</div><div><br></div><div><br></div><div><br><div class="gmail_quote">
2011/4/24 Fraser Redmond <span dir="ltr"><<a href="mailto:fraserredmond@gmail.com">fraserredmond@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Thanks David - it's more than one gateway though (different companies, different countries), so unlikely that both changed things at around the same time.<br><br>Because it was different gateways (and not all gateways), and different softphones (in different locations), the only common denominator was freeswitch.<br>
<br>Only one other thing is consistent: the gateway that works is geographically close to our server (both in California), but the 2 gateways that don't work are both overseas. Otherwise the only common link is Freeswitch.<br>
<br clear="all">Cheers,<br><font color="#888888">Fraser</font><div><div></div><div class="h5"><br><br><br>
<br><br><div class="gmail_quote">On Fri, Apr 22, 2011 at 4:26 AM, David Ponzone <span dir="ltr"><<a href="mailto:david.ponzone@ipeva.fr" target="_blank">david.ponzone@ipeva.fr</a>></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">Any possibility that the faulty gateways have changed their architecture to hide media relays behind NAT, and that you have rtp-auto-adjust disabled or not correctly configured on your SIP profile ?<div>
You may see that in the logs.</div><div>Another way to check this is to look the IP advertised by the gateway in the SDP, and to see if your FS is trying to send traffic to that IP.</div><div><br></div><div><div>
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<br><div><div>Le 21/04/2011 à 03:16, Fraser Redmond a écrit :</div><div><div></div><div><br><blockquote type="cite">But it's odd that it's only happening with some gateways, but other gateways get two-way audio fine. That's what really has me puzzled.<br>
<br>It's definitely not a problem at the extension side, as it happens with multiple phones, and multiple locations. (So not mic, firewall, router)<br>
<br clear="all">Cheers,<br>Fraser<br><br><br>
<br><br><div class="gmail_quote">On Wed, Apr 20, 2011 at 7:42 PM, Sergey Okhapkin <span dir="ltr"><<a href="mailto:sos@sokhapkin.dyndns.org" target="_blank">sos@sokhapkin.dyndns.org</a>></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">
No outgoing audio? Usually this happens if SIP ALG is enabled in your router.<br>
<div><div></div><div><br>
On Wednesday 20 April 2011, Fraser Redmond wrote:<br>
> No, I tried turning off the firewall, and as I said in the OP it works with<br>
> one of our other gateways.<br>
><br>
> Mic works on that one gateway, and during the calls where the audio isn't<br>
> transmitted the mic-indicator goes up and down.<br>
><br>
> Cheers,<br>
> Fraser<br>
><br>
> On Wed, Apr 20, 2011 at 5:46 PM, Brian West <<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>> wrote:<br>
> > firewall issue? mic doesn't work? can you get a pcap of all traffic?<br>
> ><br>
> > /b<br>
> ><br>
> > On Apr 20, 2011, at 3:04 PM, Fraser Redmond wrote:<br>
> ><br>
> > Thanks Brian, I'd appreciate you looking - I don't know what to look for<br>
> > in the sip traces (could be worth documenting some pointers in the<br>
> > wiki?)<br>
> ><br>
> > The sip trace is here:<br>
> > <a href="http://pastebin.freeswitch.org/16136" target="_blank">http://pastebin.freeswitch.org/16136</a><br>
> ><br>
> > I pressed enter a few times in the console before and after it connected<br>
> > to the extension, so about lines 490-600 is the relevant part.<br>
> ><br>
> > I also captured a pcap, in case that is of interest - let me know and<br>
> > I'll email it directly.<br>
> ><br>
> > Thanks,<br>
> > Fraser<br>
> ><br>
> ><br>
> ><br>
> > _______________________________________________<br>
> > FreeSWITCH-users mailing list<br>
> > <a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
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