[sr-dev] [kamailio/kamailio] Support for 192.0.0.0/29 as NATed source (#1488)

[Sergiu Pojoga]

It's been a while, stumbled on this case just now. It seems like either nat_uac_test() or more likely fix_nated_sdp() doesn't catch the 192.0.0.0/29 subnet as private.

**Script logic:**
route[NATMANAGE] {
...
if (nat_uac_test("8"))
   fix_nated_sdp("15");
...
}

**Sipdump:**
U 2020/04/07 10:36:02.572802 135.19.155.163:17669 -> 65.39.1.1:5060
INVITE sip:8007777777 at client.mydomain.net:5060 SIP/2.0
Via: SIP/2.0/UDP 192.0.0.254:11198;branch=z9hG4bK1066155396
From: "ME" <sip:2213 at client.mydomain.net:5060>;tag=3901872054
To: <sip:8007777777 at client.mydomain.net:5060>
Call-ID: 6_327133011 at 192.168.0.78
CSeq: 1 INVITE
Contact: <sip:2213 at 192.0.0.254:11198>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T46S 66.84.0.10
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 305

v=0
o=- 22668 22668 IN IP4 192.168.0.78
s=SDP data
c=IN IP4 192.0.0.254
t=0 0
m=audio 22936 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

U 2020/04/07 10:36:02.573384 65.39.1.1:5060 -> 135.19.155.163:17669
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.0.0.254:11198;branch=z9hG4bK1066155396;rport=17669;received=135.19.155.163
From: "ME" <sip:2213 at client.mydomain.net:5060>;tag=3901872054
To: <sip:8007777777 at client.mydomain.net:5060>
Call-ID: 6_327133011 at 192.168.0.78
CSeq: 1 INVITE
Server: NXO
Content-Length: 0

U 2020/04/07 10:36:02.573608 65.39.1.1:5060 -> 66.199.2.2:5060
INVITE sip:8007777777 at client.mydomain.net:5060 SIP/2.0
Record-Route: <sip:65.39.1.1;lr;did=faf.517>
Via: SIP/2.0/UDP 65.39.1.1;branch=z9hG4bK6ef8.b9248afe5375fc379eb33718de1f7481.0
Via: SIP/2.0/UDP 192.0.0.254:11198;rport=17669;received=135.19.155.163;branch=z9hG4bK1066155396
From: "ME" <sip:2213 at client.mydomain.net:5060>;tag=3901872054
To: <sip:8007777777 at client.mydomain.net:5060>
Call-ID: 6_327133011 at 192.168.0.78
CSeq: 1 INVITE
Contact: <sip:2213 at 192.0.0.254:11198;alias=135.19.155.163~17669~1>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 69
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 281
..
v=0.
o=- 22668 22668 IN IP4 192.168.0.78
s=SDP data.
c=IN IP4 192.0.0.254
t=0 0
m=audio 22936 RTP/AVP 9 0 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

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