[Freeswitch-users] No audio/dtmf from softphone behind NAT

[Fraser Redmond]

Thanks Brian. Sorry, should have done a full sip trace before, but here is
one now: <http://pastebin.freeswitch.org/12632>

Calling an IVR dialplan:
http://pastebin.freeswitch.org/12634

Calling from one extn to another.
http://pastebin.freeswitch.org/12633
(With this one, the source/calling softphone gets a message on it saying put
on hold by the other user - not sure if that helps.)

For what it's worth, at a couple of points when I was running the trace I
was pressing keys to generate dtmf, and nothing changed on the screen - no
activity at all.

Also, I've been able to remote desktop into a computer on another network,
and install x-lite and it can connect to our internal server and works fine,
but it can't do dtmf on the EC2 server either (so it's definitely a problem
on the server end somehow, not my local network's NAT.)

Cheers,
Fraser




On Mon, Apr 5, 2010 at 5:21 PM, Brian West <brian at freeswitch.org> wrote:

> Most likely its in the RTP stream as RFC2833 which is the reason you're not
> getting anything plus I need a FULL sip trace not this abbreviated trace.
>
> /b
>
> On Apr 5, 2010, at 11:12 AM, Fraser Redmond wrote:
>
> I've taken another stab at this one way audio problem today.
>
> 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.
>
> 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.
>
> Any suggestions of what I should try now?
>
> Would the dtmf's be sent in the sip packets, or in the rtp?
>
> To preempt the easy answers and save some time:
> - ports are open on EC2 config,
> - iptables turned off for the test,
> - RTP port range uncommented in switch.conf.xml,
> - softphone stun was set to stun.freeswitch.org
>
> Cheers,
> Fraser
>
>
>
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