NAT SIP ALG

[Eugene Grosbein]

On 13.05.2016 20:32, Larry Rosenman wrote:
> On 2016-05-13 07:30, Eugene Grosbein wrote:
>> 13.05.2016 5:26, James Lodge wrote:
>>
>>> I use siproxd on pfsense and it works really well.
>>
>> How do you use them?
>>
>> I have SIP PBX inside private network that has old slow Cisco router as 
>> NAT box.
>>
>> The PBX registers with several distinct external SIP providers and
>> accepts external
>> calls for local call center. I need to replace Cisco router with
>> something having much
>> more horsepower and consider FreeBSD/amd64 but need to make sure SIP
>> won't break.
>>
>> Will siproxd help me in such case? I still need the PBX to be able to 
>> register
>> directly to external SIP providers as it is administrated by outsourced 
>> company.
>>
> 
> My asterisk server (pbx) registers through my NAT without ANY rules or 
> what have you.  I do have the NAT
> option turned on for the 4 providers I connect to.
> 
> That said, sipoxd/pfSense may work as well.  I do go through pfSense, 
> but again do NOT have any SUP proxying.  The asterisk server just keeps 
> a registration open.
> 
> I can get and receive calls.

This network uses Asterisk too as internal IP PBX.

My router has two external internet connections supplied by distinct ISPs.
Internal PBX makes outgoing call using external SIP provider and uses something
like this in the first INVITE packet:

Content-Type: application/sdp.
Content-Length: 268.
.
v=0.
o=root 653657088 653657089 IN IP4 172.19.8.27.
s=SimplePBX.
c=IN IP4 172.19.8.27.

Intelligent NAT ALG should replace noted internal IP
with one of its external addresses corresponding to used IP route to SIP provider.
How can incoming voice RTP traffic pass without such translation?

Eugene Grosbein