I've taken another stab at this one way audio problem today.<br><br>I've run a wireshark capture, and looking at the RTP analysis it only has the down-stream, it doesn't record anything being sent upstream at all.<br>
<br>Below is the SIP graph, which shows RTP coming down, but none going up. But I don't know enough about SIP to know whether something is missing.<br><br>Any suggestions of what I should try now?<br><br>Would the dtmf's be sent in the sip packets, or in the rtp?<br>
<br>To preempt the easy answers and save some time:<br>- ports are open on EC2 config, <br>- iptables turned off for the test,<br>- RTP port range uncommented in switch.conf.xml,<br>- softphone stun was set to <a href="http://stun.freeswitch.org" target="_blank">stun.freeswitch.org</a><br>
<br clear="all">Cheers,<br>Fraser<br><br><br>|Time | 192.168.1.8 |<br>| | | 184.73.226.197 | <br>|6.488 | INVITE SDP ( BV32 BV32-FEC SPEEX SPEEX-FEC g71...iLBC g711A g) |SIP From: <a href="mailto:sip%3A12610@184.73.226.197" target="_blank">sip:12610@184.73.226.197</a> <a href="mailto:To%3Asip%3A12605@184.73.226.197" target="_blank">To:sip:12605@184.73.226.197</a><br>
| |(25829) ------------------> (5060) |<br>|6.615 | 100 Trying| |SIP Status<br>| |(25829) <------------------ (5060) |<br>|6.623 | 407 Proxy Authentication Required |SIP Status<br>
| |(25829) <------------------ (5060) |<br>|6.623 | ACK | |SIP Request<br>| |(25829) ------------------> (5060) |<br>|6.738 | INVITE SDP ( BV32 BV32-FEC SPEEX SPEEX-FEC g71...iLBC g711A g) |SIP From: <a href="mailto:sip%3A12610@184.73.226.197" target="_blank">sip:12610@184.73.226.197</a> <a href="mailto:To%3Asip%3A12605@184.73.226.197" target="_blank">To:sip:12605@184.73.226.197</a><br>
| |(25829) ------------------> (5060) |<br>|6.869 | 100 Trying| |SIP Status<br>| |(25829) <------------------ (5060) |<br>|7.070 | 183 Session Progress SDP ( g711U telephone-eve... |SIP Status<br>
| |(25829) <------------------ (5060) |<br>|7.264 | RTP (g711U) |RTP Num packets:520 Duration:10.793s SSRC:0x5433093E<br>| |(44172) <------------------ (30432) |<br>
|18.090 | 200 OK SDP ( g711U telephone-event) |SIP Status<br>| |(25829) <------------------ (5060) |<br>|18.112 | ACK | |SIP Request<br>| |(25829) ------------------> (5060) |<br>
|31.750 | BYE | |SIP Request<br>| |(25829) ------------------> (5060) |<br>|31.872 | 200 OK | |SIP Status<br>| |(25829) <------------------ (5060) |<br>
<br><br><br><br>
<br><br><div class="gmail_quote">On Sat, Apr 3, 2010 at 4:49 PM, Fraser Redmond <span dir="ltr"><<a href="mailto:fraserredmond@gmail.com" target="_blank">fraserredmond@gmail.com</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;">
I've got a FreeSwitch server up on Amazon EC2, ports wide open for my office external-IP, server iptables disabled, and changed the FreeSwitch ACL domains to "allow", so it's all wide open for now.<br><br>
In the office I'm trying to connect to the server from Bria/X-lite. I've entered a stun server (<a href="http://stun.freeswitch.org" target="_blank">stun.freeswitch.org</a>) and I can now call to the server, but not from the server. I read this page:<br>
<a href="http://wiki.freeswitch.org/wiki/Nat_stun_debug_irc" target="_blank">http://wiki.freeswitch.org/wiki/Nat_stun_debug_irc</a><br>which suggested adding this variable to the user config:<br> <a href="http://wiki.freeswitch.org/wiki/Natted_Softphone_ATA#NDLB-connectile-dysfunction" target="_blank">http://wiki.freeswitch.org/wiki/Natted_Softphone_ATA#NDLB-connectile-dysfunction</a><br>
<br>With that on I can now call to and from the server. However with or without that although I can hear audio from the server, audio to the server isn't arriving (doesn't appear in recordings), and dtmf doesn't get received either. <br>
<br>When I hang up from the client, I see in the CLI that it gets that instruction, so it hasn't started the call and lost all contact with the softphone, it's receiving some instructions, but not the audio and dtmf.<br>
<br>The problem is that both the server and client are each behind NAT, so either could be having the problem (on EC2 the auto-NAT doesn't work, so I've specified the external rtp and sip ip's.. I've also turned on aggressive-NAT in case that helps. Also I'm connecting to the server by a sub-domain (A-name) rather than IP.)<br>
<br>
I've got almost the same setup working fine on the internal network (same
dialplan and directory, and all the config is the same if it can be), so
it's got to be something to do with the NAT's.<br>
<br>Any suggestions on what the problem might be, or how to find it?<br><br clear="all">Cheers,<br><font color="#888888">Fraser<br><br><br>
</font></blockquote></div><br>