Iqbal,
Excellent summary! There are many good NAT whitepapers on the net. Here is
one that is short and clear that elaborates on your summary:
Heres my understanding...hope it helps (may even
confuse), and looking
for corrections...its a long read
1. There are 4 different types of NAT
a) Full Cone
b) restricted cone
c) port restricted
d) symmetric
a,b,c are also referred to as asymmetric NAT.
2. SIP has a problem because the siganalling uses one path and the
media stream another.
3. Full cone - Anyone of the internet can send packets to the IP:port
combo, this is mapped to a internal IP:port.
4. Restricted Cone - Only those external PC which have been contacted
are ALLOWED to contact via the mapping, i.e if I contact PC(a) from
internal Ip 10.1.1.1:123 then PC(a) can contact me on that NAT
mapping, PC(b) cannot
10.1.1.1:123 ---NAT ---> 202.70.65.78:10000 ------pc(a)
If pc(b) sends to 202.70.65.78:10000 there will be nothing sent
through to 10.1.1.1:123
5. Port restricted Cone - Same as restricted but instead of just
detecting that the source IP is that of pc(a), it also looks to see if
the source port is the same
10.1.1.1:123 --->NAT---->202.70.65.78:10000 -----> pc(a)
[213.123.324.34:8000]
The nat will only accept inbound from 213.123.324.34 and if it comes
from port 8000
6. Symmetric - This is the easy one,
10.1.1.1:1000 ----NAT -----> 200.123.123.34:1234 ----pc(a)
10.1.1.1:1000 ----NAT -----> 200.123.123.34:2222----pc(b)
In the NAT the IP:port pair is different for each external client, so
eeach external client has a mapping of its own
7. The problem RTP
In RTP the message body has the info needed for the UA to communicate
successfully. This body is called the SDP message. The problem is that
the client doesnt know anything about the NAT, hence the IP addresses
which are contained in the SDP are usually the internal ones, i.e what
the client knows....so when the endpoints want to "talk" they look at
the IP in the message and you get nothing because these are usually
internal IP addresses.
EG
INVITE sip:040600@192.168.20.2:5060 SIP/2.0.
Record-Route: <sip:143.248.130.35;ftag=3a7ceb24a6ac50c4;lr=on>.
Via: SIP/2.0/UDP 143.248.130.35;branch=z9hG4bK758e.976609c7.0.
Via: SIP/2.0/UDP
192.168.20.3;rport=1024;received=223.178.140.109;branch=z9hG4bK34efcab2403aa20d.
From: "Iqbal" <sip:040618@sip.dom.com>;tag=3a7ceb24a6ac50c4.
To: <sip:040600@sip.dom.com>.
Contact: <sip:040618@223.178.140.109:1024>.
Supported: replaces.
Call-ID: 7f2c327896a5b0e1(a)192.168.20.3.
CSeq: 8717 INVITE.
User-Agent: Grandstream HT487 1.0.5.18.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Type: application/sdp.
Content-Length: 343.
.
v=0.
o=040618 8000 1 IN IP4 192.168.20.3.
s=SIP Call.
c=IN IP4 192.168.20.3.
t=0 0.
m=audio 38660 RTP/AVP 0 8 4 18 2 15 99.
a=sendrecv.
a=rtpmap:0 PCMU/8000/3.
a=rtpmap:8 PCMA/8000/3.
This header is just like email headers hence u read it backwards, so
if you look at the line above the From: line you see the first Via,
which is what the client thinks it is i.e 192.168.20.3, BUT the proxy
is clever, it knows where it received the message from , and it adds
the rport and received tags
Via: SIP/2.0/UDP
192.168.20.3;rport=1024;received=223.178.140.109;branch=z9hG4bK34efcab2403aa20d.
Soooo the proxy can talk SIP fine, because it knows these IP
addresses.
BUT...poor old RTP is stuck because its headers or should I say
direction is held lower down :
v=0.
o=040618 8000 1 IN IP4 192.168.20.3.
s=SIP Call.
c=IN IP4 192.168.20.3.
t=0 0.
m=audio 38660 RTP/AVP 0 8 4 18 2 15 99.
a=sendrecv.
a=rtpmap:0 PCMU/8000/3.
a=rtpmap:8 PCMA/8000/3.
The client expects to receive on port m=38660 and IP c= 192.168.20.3,
which is where the other endpoint will try to send "voice" to.
8 Solution - You need to tell the client, not to act so dumb, and work
out what the NAT settings are and put them in the SDP section of the
message. So the client can ask the NAT....or it can ask someone on the
outside what it should be.
9. Ask the NAT - use UPnP...I dont have much info on this..:-)
10. Ask someone on the outside --
You send a probe packet to the server sitting outside, it then sends a
message back, with the details it received, the client then decides if
its behind a NAT. This can be used for all 4 cases of NAT.
EG lets say we send out a packet from 10.1.1.1:1000 so in the SDP
message m=1000 and c=10.1.1.1, but if I send out a probe first, and I
get back 212.134.123.23:12345 then I can rewrite the SDP so m=12345
and c=10.1.1.1 , simple
Problem -- Since NAT settings are dynamic, and hence tend to change,
you really need to get the SIP message out very soon after sending the
probing message out,
The client send and receive ports must be the same
And....if you recall the restricted cones (port restricted included)
will not allow replies unless there has been a packet sent out to that
destination first, hence the client needs to send a packet out to the
endpoint, b4 he can be allowed in via the NAT (but we dont need to
worry about that)
IT WILL NOT WORK FOR SYMMETRIC NAT....because it wont :-), because the
external UA IP:port is different to that where we sent the probe,
hence the voice packet coming back will not have the correct path set.
11. The above is usually done with a STUN server, and sending the
packet out to this server
12. Symmetric NAT--- use a relay in the middle...Nathelper+ rtpproxy
or mediaproxy
13 Nathelper
a) fix_nated_contact - rewrites contact Hf to source IP:port
b) fix_nated_sdp - rewritres media IP and also direction ????
c) force_rtp_proxy - forces media to go through proxy
d) nat_uac_test - mode=1 then as shown above the "received" header is
compared to c= in sdp
mode=2 then the Contact header is looked at to see if its private
mode=3 (1+2) means it does both of the above
14) mediaproxy is much the same...and good examples are available
15)Summary
4 types of NAT, which can be combined into 2 main sets, asymmetric and
symmetric.
Asymmetric issues can be resolved by using STUN, hence no need for
mediaproxy/nathelper
Symmetric clients cant use STUN, hence u need to use
mediaproxy/nathelper (or some other server end...also known as
far-end nat traversal solution)
However if you didnt want to use STUN and mediaproxy/nathelper, then
you could just use mediaproxy/nathelper and setup port forwarding on
your NAT device.
I think that covers most, any suggestions let me know, I wrote this
for my own use, but if its useful, I'll tidy it up.
One question....
Is there any way via a debug log, albeit, ngrep, tcpdump, sip_scenario
display that I can from the server side detect if the NAT is
asymmetric or Symmetric.
tks
Iqbal
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