[Serusers] NAT with SER and ASTERISK, strange behaviour

[Dave Bath]

 Have you enabled symmetric nat in the Cisco phone?  This is under the
setting "NAT == YES"?

Dave

-----Original Message-----
From: serusers-bounces at iptel.org [mailto:serusers-bounces at lists.iptel.org] On
Behalf Of Morten Kuehl
Sent: 14 September 2004 10:50
To: serusers at lists.iptel.org
Subject: RE: [Serusers] NAT with SER and ASTERISK, strange behaviour

Hi,

I am aware of the difference between media and signalling as well as the
way an rtp proxy works. But still do I think that this is a different
case:

When using asterisk, xlite does symetric rtp streams aka. sends from
port
8000 and listens on port 8000. The port is therefore open on nat and udp
pakets can travel through the nat with the correct public ip in the sip
messages.
When I do have the setup with ser and the sip messages look the same,
aka.
xlite says "I am listening on 8000 on public ip" and send via 8000 to
the
cisco phone which says "I am listening on port x on my ip". So there
should be no need for an rtp proxy as the rtp stream from xlite is
symetric and the nat port is open. But still there is no incoming audio.
I
can understand that with clients like MS messenger there is a need to
rewrite the sdp and sip messages.

So where is the problem in my szenario, what little magic does asterisk
do
that I do not see in the sip messages???

Cheers
Morten



> 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


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