asterisk users - Sep 2003 - SIP client<->NAT<->Asterisk<->NAT<->SIP client. only works with canreinvite=no.

Hi!

I have this configuration:

SIP client A <-> NAT box A (real external IP) <-> Asterisk server
(real
IP) <-> (real external IP) NAT box B <-> SIP client B

The echo test form any of the clients to the asterisk server is working 
just fine, even without canreinvite=no.

When I try to call from SIP client A to B, wihtout the canreinvite=no in 
the sip.conf, the call doesn't even ring.

Then I add the canreinvite=no to BOTH clients on the sip.conf, it starts 
to work.  The problem is that all voice data goes through my asterisk 
server, so the delay is longer.

Also, this config doesn't work:

SIP client A <-> NAT box A (real external IP, only one) <-> Asterisk
server (real IP)
SIP client C <-> NAT box A (real external IP, only one) <-> Asterisk
server (real IP).

When I try to call from A to C or C to A, the phone doesn't even ring, 
again, the echo test work just fine. 

SIP client A and SIP client C are in the same LAN, and both goes through 
NAT box A to the same asterisk server.

In the case of clients A and C, the native bridge would be great, 
because it would save bandwith to both, my client, and me, and the voice 
delay would be almost nothing.

My problem is: According to the data I got from the sip debug and the 
X-lite debug outputs, I don't see any reazon why the native bridge can't
work, both clients gets different ports on the outside IP of the nat 
box, and that port is correctly recognized, and the reinvite packet is 
correctly sent.

Can anybody explain me what does canreinvite=yes really does?

Any ideas on the client A to C (same LAN, same NAT box, unique outside 
IP, same * server)?

Thanks in advance,

Sincerely,

Ildefonso Camargo
icamargo@unet.edu.ve

> Can anybody explain me what does canreinvite=yes really does?
> 
Not sure how technical an answer you want becasue it look slike you know whats
going on but as I unterstand it "canreinvite=no" tells the UA that
reinvite is not supported and so causes all the RTP traffic to be routed via the
* server.. I played with many nat settings and port forwarding settings and it
ended up that "canreinvite=no" was the solution to my problems as
well.. the downside is that it requires more bandwidth at the central site but
the plus side is that it works through NAT..

> Any ideas on the client A to C (same LAN, same NAT box, unique outside 
> IP, same * server)?
> 
Only thing that springs to mind is to install another * box internally and then
use IAX to connect the internal * box to the external one.. then the internal
phone will call each other without crossing the NAT..

Later..
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I have been trying to get SIP UA work with NAT but i have no been
successful has any one got  NATed ATA working(i.e an ATA witha private IP
working with NAT).
Asterisk registers the 192.168.0.3 Ip but no call go through at all,
infact there is no log of any call made on asterisk console.

can anyone please send me the sip.conf and ATA 186 configs of a NATed ATA
to working with *.
This what i have in my sip.conf

[2222]
type=friend
username=2222
transfer=yes
nat=yes
canreinvite=no
context=myata
host=dynamic
permit=0.0.0.0/0.0.0.0
accountcode=mi100

  ATA configs
IP=192.168.0.3
staticRoute=192.168.0.2
mask=255.255.255.0
dhcp=0
GkorProxy= (*'s public IP)
gateway= (*'s Public IP)
outbound Proxy=(*'s public IP)
NATIP= (host machine's Public IP)



On Thu, 11 Sep 2003, Jose Ildefonso Camargo Tolosa wrote:
> Hi!
>
> I have this configuration:
>
> SIP client A <-> NAT box A (real external IP) <-> Asterisk
server (real
> IP) <-> (real external IP) NAT box B <-> SIP client B
>
> The echo test form any of the clients to the asterisk server is working
> just fine, even without canreinvite=no.
>
> When I try to call from SIP client A to B, wihtout the canreinvite=no in
> the sip.conf, the call doesn't even ring.
>
> Then I add the canreinvite=no to BOTH clients on the sip.conf, it starts
> to work.  The problem is that all voice data goes through my asterisk
> server, so the delay is longer.
>
> Also, this config doesn't work:
>
> SIP client A <-> NAT box A (real external IP, only one) <->
Asterisk
> server (real IP)
> SIP client C <-> NAT box A (real external IP, only one) <->
Asterisk
> server (real IP).
>
> When I try to call from A to C or C to A, the phone doesn't even ring,
> again, the echo test work just fine.
>
> SIP client A and SIP client C are in the same LAN, and both goes through
> NAT box A to the same asterisk server.
>
> In the case of clients A and C, the native bridge would be great,
> because it would save bandwith to both, my client, and me, and the voice
> delay would be almost nothing.
>
> My problem is: According to the data I got from the sip debug and the
> X-lite debug outputs, I don't see any reazon why the native bridge
can't
> work, both clients gets different ports on the outside IP of the nat
> box, and that port is correctly recognized, and the reinvite packet is
> correctly sent.
>
> Can anybody explain me what does canreinvite=yes really does?
>
> Any ideas on the client A to C (same LAN, same NAT box, unique outside
> IP, same * server)?
>
> Thanks in advance,
>
> Sincerely,
>
> Ildefonso Camargo
> icamargo@unet.edu.ve
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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