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

Iqbal iqbal at gigo.co.uk
Mon Apr 4 13:31:49 CEST 2005 

Okay so in order to get is straight

1. You can use just SER to deal with all types of NAT, this means that 
mediaproxy will get hit
2. You can use STUN, to get rid of clients which are asymmetric (is 
there alist of these...in fact will post with new topic)
3. Or you can ensure none of ur clients use NAT :-)

Iqbal

Greger V. Teigre wrote:

> Ask Paul and he'll say yes, but he mostly supports ATAs with public 
> IPs and only occasionally meet an IP phone behind a NAT.  I actually 
> use STUN, but I have tested the behavior for each client (and each 
> firmware).  This reduces the number of proxied sessions to those where 
> a symmetric NAT is involved.
> g-)
>
> Iqbal wrote:
>
>> Thanks Greg, so from what I understand its better NOT to use STUN,
>> and do fat end traversal all the time, i.e do it at SER level whether
>> its asymmetric or symmetric
>>
>> Iqbal
>>
>> On 4/3/2005, "Greger V. Teigre" <greger at teigre.com> wrote:
>>
>>> Iqbal wrote:
>>>
>>>> maybe well written but I have come across another question, in the
>>>> ser.cfg file on onsip it mentions that with correct use of the
>>>> mediaproxy you can deal with asymmetric and symmetric, hence my
>>>> summary may be wrong.
>>>
>>>
>>> No, your summary is correct.  The config files in the onsip.org
>>> document have the assumptions that STUN is NOT used.  Hence, all
>>> NATed devices will be proxied.  When STUN is used, it is transparent
>>> and the config files will see the client as non-NATed.
>>>
>>>> Symmetric CANNOT go through STUN, it wont work. hence the STUN is no
>>>> use, and yes then ser will see it as natted.
>>>
>>>
>>> One big problem is that many clients have faulty STUN
>>> implementations, even specific firmware can have problems
>>> (Grandstream changed its behavior to wrong handling of symmetric NAT
>>> in 10.5.16, I think).
>>>
>>> In fact, it is possible both to detect (some of) these faulty
>>> implementations (i.e. faulty handling of symmetric NAT, ex. using
>>> test 16 in nathelper), but not if ex. a full-cone is detected when
>>> in fact, it's a restricted NAT. (Let me add that there also are many
>>> mutations/implemenations of the four NAT types that can confuse
>>> STUN...)
>>>
>>> Conclusion: ALWAYS test the clients you want to use with STUN!
>>>
>>> We are considering adding an appendix with advanced NAT techniques in
>>> ser.cfg including on where you can use STUN (test your clients) as
>>> well as direction=active supported PSTN gateways (used to allow
>>> non-proxied outgoing calls from a call behind a symmetric NAT) to
>>> reduce the proxied calls to the following:
>>> - SIP-SIP when one or both are behind symmetric
>>> - PSTN-SIP when the SIP client is behind symmetric
>>>
>>> We'll see when we get the time. There are lots of more basic stuff
>>> we need to do first.
>>> g-)
>>>
>>>> If asymmetric comes via STUN, and the client understands what the
>>>> STUN server has sent back, and can fix itself then ser will NOT see
>>>> it has natted, since its been fixed
>>>>
>>>> Iqbal
>>>>
>>>> On 4/2/2005, "Mohammad Khan" <info at beeplove.com> wrote:
>>>>
>>>>> Very nice and well written, Many many thanks.
>>>>>
>>>>> One question,
>>>>> If a client from Assymetric NAT and it comes through STUN, does SER
>>>>> see that client as a *nated* client?
>>>>> And,
>>>>> if a client from Symentric NAT and it still comes through STUN, SER
>>>>> see that client as Nated client, right?
>>>>>
>>>>>
>>>>> See below:
>>>>>
>>>>>
>>>>>
>>>>> Iqbal wrote:
>>>>>
>>>>>> 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)
>>>>>
>>>>>            ^^
>>>>> you mean 10.1.1.x ( x is other than 1), right?
>>>>> I think, this was a typo.
>>>>>
>>>>> Thanks,
>>>>> Mohammad
>>>>>
>>>>>
>>>>>> 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 at 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 at sip.dom.com>;tag=3a7ceb24a6ac50c4.
>>>>>> To: <sip:040600 at sip.dom.com>.
>>>>>> Contact: <sip:040618 at 223.178.140.109:1024>.
>>>>>> Supported: replaces.
>>>>>> Call-ID: 7f2c327896a5b0e1 at 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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>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>
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>
>
> .
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