[Serusers] NAT with SER and ASTERISK, strange behaviour

[Zeus Ng]

Now, symmetric rtp only means your ua will send and receive rtp packets on
the same port, 8000 in your case. It does not automatically means that the
NAT device will open port 8000 for incoming packets, unless you have port
forwarding in place. Mostly importantly, your NAT device does not necessary
use port 8000 to map your port 8000 for outgoing. It can be something else,
say 60000.

For the Cisco side, it will see rtp stream coming from <ua1 pub ip:60000>.
>From the SDP it received previously, it will send rtp stream back to <ua1
pub ip:8000>. Now, unless you have port forwarding in place, your NAT device
will drop the packets as the "hole" is not opened. 

Both Asterisk and rtpproxy do this by sending the packet back to where it
receive from, that is <ua1 pub ip:60000>, and ignoring the SDP. Asterisk
does this automatically and you have to instruct SER and rtpproxy to do that
manually.

Hope it answer your question.

Zeus


> -----Original Message-----
> From: serusers-bounces at lists.iptel.org 
> [mailto:serusers-bounces at lists.iptel.org] On Behalf Of Morten Kuehl
> Sent: Tuesday, 14 September 2004 7:50 PM
> 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
> 
> 
> _______________________________________________
> Serusers mailing list
> serusers at lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
>