One way audio usually is a result of NAT.<div><br></div><div>Not all NAT routers operate in the same way, and not all softphones handle it in the same way, so that may explain why it works in one location but not the other. I assume the softphones were behind NAT?</div>
<div><br></div><div>Set the FS log level to debug, and check the IP in the remote SDP. You need the phone's external IP in there. Phones usually determine that using STUN. Some NAT routers have a feature called SIP ALG to rewrite it for phones not doing it themselves, but that often will conflict with phones that have already put the correct IP (and also cannot work with TLS).</div>
<div><br></div><div>In short check where RTP is being sent to is correct...</div><div><br></div><div>Without the correct IP FS also has a last-resort feature where it detects the IP the caller sends audio from and replies to there (RTP auto-adjust). A log entry indicates when that happens. But that requires FS to receive audio, which means it can't work until the caller starts speaking after the call is answered (so might miss hearing the callee say 'hello'), and the caller will never hear ringing (unless generated locally by the phone when it gets the SIP 183/180 response).</div>
<div><br></div><div>The NAT page on the Wiki may be informative: <a href="http://wiki.freeswitch.org/wiki/NAT">http://wiki.freeswitch.org/wiki/NAT</a></div><div><br></div><div>Just to complicate things Amazon EC2 runs behind a NAT layer too AFAIK. I don't have any personal experience with EC2.</div>
<div><br></div><div>-Steve</div><div><br></div><div><br></div><div><br><div><div><div class="gmail_quote">On 28 January 2013 09:26, Stuart Mills <span dir="ltr"><<a href="mailto:stuart.mills3@btopenworld.com" target="_blank">stuart.mills3@btopenworld.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">
<div dir="ltr">
<div style="font-size:12pt;font-family:'Calibri'">
<div>Hi,</div>
<div> </div>
<div>I have a strange problem using FreeSwitch on an Amazon EC2 instance and was
wondering if anyone out there could help me.</div>
<div> </div>
<div>The problem seems to be that I can register an extension from a remote
network in say France, phone works ok but when passing calls to it (from sip
peer through JavaScript to user extension) I get one way audio, the caller can
hear me on the softphone but I can’t hear them – like the reverse audio path
can’t be established.</div>
<div> </div>
<div>That would be ok and fixable if the issue was consistent but on another
phone registered from within the UK, it all works fine and I get a good 2 way
conversation.</div>
<div> </div>
<div>If I dial from one softphone in France directly to the other in the UK,
that works too.</div>
<div> </div>
<div>I followed the wiki’s guide to installing on Amazon and have everything
right, otherwise I doubt it would work at all.</div>
<div><br>Is anyone able to help me or have experience of this type of issue?
Maybe it’s a local router issue to the phone in France but I didn’t need to do
anything special to the one in the UK to work, so not sure it could be that
either.</div>
<div> </div>
<div>Kind Regards,</div><span class="HOEnZb"><font color="#888888">
<div> </div>
<div>Stuart</div></font></span></div></div></div>
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