[Serusers] NAT with SER and ASTERISK, strange behaviour

Dave Bath dave at fuuz.com
Tue Sep 14 09:14:22 CEST 2004


SER is simply a proxy - it does not handle media in the same way that
asterisk does.  When you have both clients registered with SER, once the
initial call set up has been completed, no further traffic runs through
SER.  Search this list for explanations as to why RTP traffic doesn't
really run through NAT without a helping hand.  

If you want to be able to make calls without any special client/NAT
router settings, check out RTPproxy/NAThelper and Mediaproxy - they do
the RTP proxy bit that Asterisk has built in.

Hope this helps.

Dave

-----Original Message-----
From: serusers-bounces at iptel.org [mailto:serusers-bounces at lists.iptel.org] On
Behalf Of Morten Kuehl
Sent: 13 September 2004 22:35
To: serusers at lists.iptel.org
Subject: [Serusers] NAT with SER and ASTERISK, strange behaviour

Hi folks,
I just spent the evening with trying to find a logical solution to a NAT
problem but I had no success. I decided therefore to go to bed and let
the
more intelligent guys have a guess ;):

I do have the following setup:
Cisco 7960G with public IP
Xlite behind router with NAT
Ser server with a public ip
Asterisk with a public ip

Xlite finds out that it is behind a symetric firewall while starting and
sends the external ip of the nat device in its sip messages.

I want to call the xlite client behind NAT from the Cisco phone. It
works
when the cisco phone and the xlite client are registered with the
asterisk
server. I have set nat=yes and canreinvite=no in the account settings
for
the xlite client in the sip.conf of the asterisk server. The Cisco phone
has a normal account. I had a look at the sip messages and as
configured,
the Asterisk server works as a rtp proxy for the media stream. Audio
works
in both ways.

When I use ser and register both clients with ser and start a call,
audio
works only outbound from xlite to the cisco phone but not inbound from
the
cisco phone to xlite through the nat device.

I digged the sip messages several times and did not see any big
difference
between the two sip conversations, besides having the asterisk as an rtp
proxy in the middle in the first attempt. Neither did I see an approach
of
asterisk to make xlite pinhole the nat device as describe for example in
the sip cookbook.

Unfortunately I am stuck at this stage and cannot find a logical
explanation for the described behaviour. If anyone can give me a hint or
needs further information to assist, please let me know!

Cheers
Morten



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