[Asterisk-Users] Sip UA behind NAT

[Eric Yu-Wei Sung]

Hi, is there any way I could make this work without having to explicitly 
perform port forwarding for RTP traffic at my NAT? (i.e. NAT 
transparently sets up the RTP channel for the internal SIP UA with the 
external SIP UA) Thanks Eric Date: Fri, 03 Jun 2005 09:13:18 -0500 From: 
Mike Holloway <asterisk-users at citrini.com> Subject: Re: [Asterisk-Users] 
Sip UA behind NAT To: Asterisk Users Mailing List - Non-Commercial 
Discussion <asterisk-users at lists.digium.com> Message-ID: 
<42A0657E.3080604 at citrini.com> Content-Type: text/plain; 
charset=ISO-8859-1; format=flowed 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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