Re: [Serusers] NAT NAT..and a little more NAT

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,

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 _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers