[Asterisk-Users] Sip UA behind NAT

[Mike Holloway]

Eric,

The problem you are seeing is because the RTP (voice) packets being sent 
towards the NAT'd UA are being blocked by the NAT router.

The UA being used behind NAT will need to have a static IP address set 
(e.g. 192.168.1.50) and on the NAT router you will need to permanently 
forward (port forward) SIP and RTP ports to the internal IP address.  I 
recommend ports 5060 and 16384-16400.  On the NAT'd UA, set the SIP port 
to 5060 and the RTP ports to 16384-16400.  If your UA only supports one 
RTP port, just use 16384.


As Forrest noted, you will also want to set canreinvite=no in sip.conf 
for the NAT'd UA. You should also set nat=yes, which will force asterisk 
to re-write SIP packets coming from the NAT'd UA to the correct external 
IP address of the NAT router.


-mike


Eric Yu-Wei Sung wrote:
> I am trying to make 1 soft SIP UA behind NAT connect to a public hard 
> CISCO UA via a public asterisk server. The CISCO UA can hear the voice 
> from the SIP UA but not vice versa. I do set nat to yes for the soft 
> phone. Any help would be greatly appreciated.
> 
> Below is my sip.conf
> 
> [general]
> 
> port = 8060           ; Port to bind to (SIP is 5060)
> bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
> allow=all             ; Allow all codecs
> context = bogon-calls ; Send SIP callers that we don't know about here
> 
> [2000]    ; soft phone behind NAT
> 
> type=friend           ; This device takes and makes calls
> username=2000         ; Username on device
> host=dynamic          ; This host is not on the same IP addr every time
> context=from-sip      ; Inbound calls from this host go here
> mailbox=100           ; Activate the message waiting light if this
>                      ; voicemailbox has messages in it
> nat=yes
> 
> [2002]                ; CISCO hard phone
> 
> type=friend
> username=2002
> secret=2002
> host=dynamic
> context=from-sip
> mailbox=103
> 
> 
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