<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html; charset=UTF-8" http-equiv="Content-Type">
</head>
<body text="#000000" bgcolor="#ffffff">
Peter and David, thanks for trying to help out.<br>
<br>
I did wonder about that too, Peter, but it seems to me that my VoIP
adapter at 192.168.11.250 sends out an RTCP receiver report right
before it ends a call. This is shown in <a
href="http://pastebin.freeswitch.org/14457">the trace of packet
exchange between AsteriskWin32 and the VoIP adapter</a>. In case
you did not see my original posting, AsteriskWin32 does do what I
want to achieve, i.e., automatically answers incoming calls to my
DID number.<br>
<br>
I have FreeSWITCH at 192.168.11.11 register with the VoIP adapter,
which is a SIP gateway provided by my SIP service provider. When I
call FS's extension number from another softphone, also on the home
LAN, which is also registered with the same VoIP adapter, FS answers
automatically as I expect it to. The RTP steam is established
between FS and the softphone directly. This part is different from
incoming calls' case, where the VoIP adapter seems to act as a proxy
and tries to establish an RTP stream between itself and FS. <br>
<br>
The VoIP adapter sends an ACK back to FS when FS accepts an INVITE
with an OK. I'd think everything is hunky dory up to that point.
What I do not get is why the VoIP adapter decides to end the call
immediately after (and sends an RTCP receiver report). I am guessing
it is because there is something wrong with the RTP communication
that is supposed to follow.<br>
<br>
I am thinking it is either because: A. the VoIP adapter somehow
couldn't send RTP packets, or B. the VoIP adapter didn't like the
RTP packets that it received from FS. I don't think A. is the case.
The firewall of the PC on which FS runs was turned off during
testing and the VoIP adapter has no problem sending RTP packets to
AsteriskWin32 on the same PC (I did not run FS and Asterisk
simultaneously). Since I could not find anything particularly odd
about RTP packets sent by FS, I have no idea if B. is a possibility.<br>
<br>
By the way, my home LAN is set up in a weird way (please see
<a class="moz-txt-link-freetext" href="http://i139.photobucket.com/albums/q302/starplatina/For%20Blogs/HomeLANSetup-1.jpg">http://i139.photobucket.com/albums/q302/starplatina/For%20Blogs/HomeLANSetup-1.jpg</a>).
Double NAT layers are understandably a concern, but since both the
VoIP adapter and FS (and the softphone too) are accessible on the
inner layer of NAT, this should not be a problem... I would think
(Correct me if I am wrong). I do not need to and thus am not trying
to access FS from the Internet.<br>
<br>
I would welcome any input. Thanks again for your help!<br>
<br>
Yasuro<br>
<br>
<br>
<br>
Peter Steinbach wrote (11/15/2010 9:01 AM):
<blockquote cite="mid:4CE07843.8040209@telefaks.de" type="cite">
<pre wrap="">Thanks David,
missed the "C".
But anyway, I am wondering why .250 sends a BYE right after reception of
"Destination unreachable (Port unreachable)".
</pre>
</blockquote>
<br>
</body>
</html>