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

[Fraser Redmond]

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


|Time     | 192.168.1.8                           |
|         |                   | 184.73.226.197    |
|6.488    |         INVITE SDP ( BV32 BV32-FEC SPEEX SPEEX-FEC g71...iLBC
g711A g)          |SIP From:
sip:12610 at 184.73.226.197<sip%3A12610 at 184.73.226.197>
To:sip:12605 at 184.73.226.197 <To%3Asip%3A12605 at 184.73.226.197>
|         |(25829)  ------------------>  (5060)   |
|6.615    |         100 Trying|                   |SIP Status
|         |(25829)  <------------------  (5060)   |
|6.623    |         407 Proxy Authentication Required          |SIP Status
|         |(25829)  <------------------  (5060)   |
|6.623    |         ACK       |                   |SIP Request
|         |(25829)  ------------------>  (5060)   |
|6.738    |         INVITE SDP ( BV32 BV32-FEC SPEEX SPEEX-FEC g71...iLBC
g711A g)          |SIP From:
sip:12610 at 184.73.226.197<sip%3A12610 at 184.73.226.197>
To:sip:12605 at 184.73.226.197 <To%3Asip%3A12605 at 184.73.226.197>
|         |(25829)  ------------------>  (5060)   |
|6.869    |         100 Trying|                   |SIP Status
|         |(25829)  <------------------  (5060)   |
|7.070    |         183 Session Progress SDP ( g711U
telephone-eve...          |SIP Status
|         |(25829)  <------------------  (5060)   |
|7.264    |         RTP (g711U)                   |RTP Num packets:520
Duration:10.793s SSRC:0x5433093E
|         |(44172)  <------------------  (30432)  |
|18.090   |         200 OK SDP ( g711U telephone-event)          |SIP Status
|         |(25829)  <------------------  (5060)   |
|18.112   |         ACK       |                   |SIP Request
|         |(25829)  ------------------>  (5060)   |
|31.750   |         BYE       |                   |SIP Request
|         |(25829)  ------------------>  (5060)   |
|31.872   |         200 OK    |                   |SIP Status
|         |(25829)  <------------------  (5060)   |






On Sat, Apr 3, 2010 at 4:49 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> 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.
>
> In the office I'm trying to connect to the server from Bria/X-lite. I've
> entered a stun server (stun.freeswitch.org) and I can now call to the
> server, but not from the server. I read this page:
>    http://wiki.freeswitch.org/wiki/Nat_stun_debug_irc
> which suggested adding this variable to the user config:
>
> http://wiki.freeswitch.org/wiki/Natted_Softphone_ATA#NDLB-connectile-dysfunction
>
> 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.
>
> 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.
>
> 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.)
>
> 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.
>
> Any suggestions on what the problem might be, or how to find it?
>
> Cheers,
> Fraser
>
>
>
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