[asterisk-users] Asterisk and Client NAT

[G B]

Hi,

I am trying to run an Asterisk (1.4.10) server on Ubuntu Linux Fiesty Fawn. The server is behind NAT. I am testing SIP with the X-Lite client from xten. The client is also behind NAT.

I realize that this is amongst the worst configurations, but I have been made to believe that it can work... eventually. However, currently SIP call set up seems to go fine, but no media is transferred in either direction. For example, the following is output on the asterisk CLI despite no voice being heard.  
-- Executing [101 at john:1] Playback('SIP/john-081da978', 'hello-world') in new stack 

I have used tcpdump, ethereal and RTP debug to trace the problem. I have the following data:

1. My SIP client is correctly receiving and processing SIP/SD packets. Ethereal indicates that the client is told to send RTP traffic to server port 50296 and all packets are sent to this port. However, this port IS NOT in the range specified in rtp.conf. This is my first point of confusion.

2. Pushing onward, I forwarded ALL ports from my router (NAT) to the Asterisk server to see if Asterisk would then pick up the voice stream. Tcp dump on the Asterisk server indicates:
02:48:30.082693 IP pool-71-107-141-25.lsanca.dsl-w.verizon.net.50943 > gaurav-desktop.local.50296: UDP, length 172

Each of these lines match a line in my ethereal capture on the client. However, still no voice! This is my second point of confusion.

3. Finally, I tried using RTP debug to trace where audio packets from the server are being sent. The output I got:

Sent RTP packet to      192.168.1.47:54626 (type 00, seq 058094, ts 012000, len 000160)

I have no idea where 192.168.1.47:54626 came from. There is no computer on my server's LAN with that local ip, and it is NOT the local IP of my client (which was 10.0.0.3).This is the third point of confusion.

My conf files are attached. Important details are below:

1. sip.conf
[global]
nat=yes
canreinvite=no

[john]
context=john
nat=yes
canreinvite=no
host=dynamic

2. extensions.conf

[john]

exten => 100,1,Dial(SIP/john) ; loopback
exten => 101,1,Playback(hello-world) ; the basics

3. rtp.conf

rtpstart=10000
rtpend=20000

Your expertise would be appreciated. My sincere thanks for your time and help in advance.

--G




_________________________________________________________________
Explore the seven wonders of the world
http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070819/4e90b4d5/attachment-0001.htm 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: sip.conf
Type: application/octet-stream
Size: 3106 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070819/4e90b4d5/attachment-0003.obj 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: extensions.conf
Type: application/octet-stream
Size: 13021 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070819/4e90b4d5/attachment-0004.obj 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: rtp.conf
Type: application/octet-stream
Size: 496 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070819/4e90b4d5/attachment-0005.obj