RE: nf_nat_sip questions

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You did not define the internal networks properly in the pbx. Asterisk needs to know which nets are internal and which are external. Typically in the sip.conf file you will find the local nets defined.

Bob Reiber
BK Sales and Service
3211 Longfellow Dr
Belmont, CA 94002
Tel: 650 376 1122
Fax: 650 240 4556

-----Original Message-----
From: netfilter-owner@xxxxxxxxxxxxxxx [mailto:netfilter-owner@xxxxxxxxxxxxxxx] On Behalf Of Rodrigo Stuffs
Sent: Tuesday, June 25, 2013 9:30 AM
To: netfilter@xxxxxxxxxxxxxxx
Subject: nf_nat_sip questions

Hello List,

Let a scenario:

[ Internal network 172.16.9.0/24 eth1 ] ---- <raspberry pi> ---- [ valid internet IP address eth0 ]

Recently, I got myself adventuring on Asterisk, and in order to address some issues, I found nf_nat_sip. So, my Raspberry Pi is doing the IP masquerading job for my internal network. And then, I installed Asterisk in the Pi, which is bound to both internal and external interfaces.

Ok, the internal softphones works just nicely. It happens that external phones have 1-way communication (just listens) with internal phones.

While running a tcpdump, I found that SIP was telling my external phones to send the RTP data to a internal IP address instead. See a dump below:

--------------------8<-------------------

INVITE sip:soft-01@179.245.20.148:5060 SIP/2.0
Via: SIP/2.0/UDP <home valid ip address>:5060;branch=z9hG4bK7a17bb7b;rport
Max-Forwards: 70
From: <sip:102@<home hostname>>;tag=as1a71f0f6
To: "soft-01" <sip:soft-01@<home hostname>:5060>;tag=63f8e4a543
Contact: <sip:102@<home valid ip address>:5060>
Call-ID: cf6587a53b2ef650
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.13.1~dfsg-3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 899324373 899324374 IN IP4 172.16.9.5   <------ WRONG!
s=Asterisk PBX 1.8.13.1~dfsg-3
c=IN IP4 172.16.9.5   <-------- WRONG!
t=0 0
m=audio 16428 RTP/AVP 0 125
a=rtpmap:0 PCMU/8000
a=rtpmap:125 telephone-event/8000
a=fmtp:125 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

--------------------8<-------------------

As I highlighted in the dump (seek for WRONG!), SIP is telling my external phone to talk back to a internal phone, so this is why the internal peer doesn't hear back.

Ok, just prior to send me to whine to asterisk-users list, here comes my NF relevant questions.

1. Is just loading the module nf_nat_sip enough, is it active and mangling packets? Or should I tie it to some fancy iptables rule?
http://people.netfilter.org/chentschel/docs/sip-conntrack-nat.html is not very clear about it.

2. Does nf_nat_sip mangles that IP address, in o= and c= lines? Does it mangle whatever else?

3. As I described, the Pi also serves internal network. Is there a risk to nf_nat_sip *also* mangle the *internal - internal*  SIP traffic to the server?

4. If 2 is true, how can I mitigate it? -j CT --notrack -o <internal if> ?

Thanks a lot,

- RF.
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