I have spent many hours working on NAT issues on client end, my server has a public address.
With CISCO brand phones I did not need any non-standards compliant settings, just turning on all the choices in the CISCO web setup NAT section. However, with Polycom 335 phones (as of Dec 2012) I could not get registered or get audio without the following:
* NDLB-connectile-dysfunction
* NDLB-force-rport
* Enable SIP ALG on my FIOS router.
With those setting however, this has worked perfectly. Also note that when I turned on SIP ALG, my Cisco phones quite working until I added the NDLB parameter/variable to the Cisco <user> in the directory. They seem to be quite complimentary but seem be requirements for each other.
I really tried to stay away from SIP ALG because so many posts were so negative about it. Without the NDLB “flags” I could never see any difference when enabling SIP ALG. The combination for me has been fantastic.
HOWEVER, since there are so many different versions of “success” in the IRC and Wiki, I am pretty sure that other router brands with different SIP ALG implementations and/or other phone brands or even firmware versions may need different configurations. It is almost like we just need a checklist that says try these combinations until you find one that fits your site.
HTH,
sean
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Cal Leeming [Simplicity Media Ltd]
Sent: Sunday, December 16, 2012 11:15 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] NAT traversal - the final say..!
Any and all feedback on this thread would be much welcomed.
Hello,
There seems to be a large number of discussions surrounding NAT traversal, as well as lots of documentation, but with no concrete answers.
The NAT related wiki documentation is tedious, and depending on the outcome of this thread, I'd like to spend some time cleaning it up.
The most common problem (the same as ours) was having a router with broken ALG and a softphone that does not seem to work with STUN.
The following REGISTER is sent from a phone.
REGISTER sip:1.2.3.4:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:57787;branch=z9hG4bK-d8754z-b31b18401713de75-1---d8754z-;rport
Max-Forwards: 70
To: "foxx"<sip:2000@1.2.3.4:5060>
From: "foxx"<sip:2000@1.2.3.4:5060>;tag=83311448
Call-ID: NGQyMjJkODlhMzQ1ZWY4ZDk4ZjZmZWRhODU0NWE5YWI.
CSeq: 7 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 6.0.25732.0
Content-Length: 0
As you can see, the client's public IP is not specified anywhere. FreeSWITCH offers several ways around this, the main ones being;
* NDLB-connectile-dysfunction
* NDLB-force-rport
* apply-nat-acl
* sip-force-contact
The one that has worked in our case was "NDLB-connectile-dysfunction" (otherwise known as NAT HACK), however there seems to be a lot of negative comments about using this.
From what I can tell, the general argument is that NAT HACK is considered a non RFC compliant hack, and the SIP phones should be doing a better job of keeping to the RFCs.
In principle, this is a fair argument - but in practise, it's not a reasonable assumption that all phones are RFC compliant, and (imho) not a reasonable argument to have this functionality disabled by default.
So, I'd like to present the following arguments;
* Are there any other negative aspects about using NDLB-connectile-dysfunction, other than it is a non compliant RFC hack?
* Why is NDLB-connectile-dysfunction not enabled by default when certain conditions are met? In the event that FreeSWITCH receives a REGISTER from a phone specifying a Contact/Via as 192.168.0.0/16, but received on a public IP, then it should be obvious that NAT is broken and automatically try to circumvent it.
* People seem to get confused between server side and client side NAT problems, and that they both need to be resolved in a different way. The documentation doesn't seem to reflect this clearly.
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