Do you see any entries in the FS logs relating to rtp auto-adjust at about the time the audio starts working?<div><br></div><div>If so that could indicate FS is getting an incorrect (internal) IP in the SDP from the client, and audio is starting when the client starts sending to FS after a few seconds.</div>
<div><br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><span style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:13.333333969116211px;background-color:rgb(255,255,255)">The other party have to hung up, for example, in </span><span style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:13.333333969116211px;background-color:rgb(255,255,255)">order for call to end.</span></blockquote>
<div><br></div><div>This also indicates a NAT issue. The BYE hangup gets sent to the IP:port in the Contact header, not the address the INVITE came from. If the contact header is incorrect the BYE goes to the wrong place and so the client never sees it.</div>
<div><br></div><div>If you can't correct that on the client end (STUN, set manually, SIP ALG etc) then try <a href="http://wiki.freeswitch.org/wiki/NDLB#NDLB-connectile-dysfunction">http://wiki.freeswitch.org/wiki/NDLB#NDLB-connectile-dysfunction</a> and/or <a href="http://wiki.freeswitch.org/wiki/NDLB#NDLB-force-rport">http://wiki.freeswitch.org/wiki/NDLB#NDLB-force-rport</a> which will tell FS to break from the SIP standard to support buggy clients.</div>
<div><br></div><div>-Steve</div><div><br><br><br><br><div class="gmail_quote">On 10 December 2012 18:22, Mimiko <span dir="ltr"><<a href="mailto:vbvbrj@gmail.com" target="_blank">vbvbrj@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>On 10.12.2012 20:00, Steven Ayre wrote:<br>
> So to clarify:<br>
><br>
> Client -> Internet -> NAT -> FS<br>
> works<br>
><br>
> Client -> NAT -> Internet -> NAT -> FS<br>
> does not?<br>
><br>
> Since the FS NAT side appears to be working (from the 1st case), I'm<br>
> guessing it's probably either a problem at the client side (with client<br>
> or NAT router) or perhaps the double-NAT is causing an issue. Personally<br>
> I haven't run FS behind NAT, so it's not a situation I've had to handle.<br>
<br>
</div>Thank you for a comprehensive response.<br>
<br>
What I've seen now. Audio actually comes up, but it is taking a while.<br>
Sometimes is ~5sec, sometimes more, sometimes never. Also, sip commands<br>
does not take action. The other party have to hung up, for example, in<br>
order for call to end.<br>
<div><div><br>
--<br>
Mimiko desu.<br>
<br>
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