Discussion: NAT and VOIP

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Re: Broken link to pdf

There is a backup at the internet archive though

http://web.archive.org/web/*/http://voip-itec.tamu.edu/files/reference/voip-nat.pdf



by urbieta, Wednesday 27 of June, 2007 (17:10:19 UTC)
warning

i alwayes get the message at the asterisk consol
Apr 14 07:28:53 WARNING[32646]: channel.c:3645 ast_channel_bridge: Can't make SIP/ss79.xipx.com-082f9078 and SIP/196.219.50.76-082fe5b8 compatible
Apr 14 07:28:53 WARNING[32646]: res_features.c:1385 ast_bridge_call: Bridge failed on channels SIP/ss79.xipx.com-082f9078 and SIP/196.219.50.76-082fe5b8



by ventora, Monday 11 of June, 2012 (05:27:00 UTC)
Re: Just port forward your listening and RTP ports

In my oppinion, any solution should try to avoid modifying the devices that give the client access to the internet (such as routers, firewalls, etc). In that way solutions are more flexible and versatile.

In that sense, using STUN could more suitable. Though not supported by all the clients, is sufficiently implemented and workable (as far as i have tested it...).

by alvarocanivell, Thursday 12 of January, 2006 (17:21:45 UTC)
Check out Edgewater Networks !

(:cool:) They have a box that handles this very well. Their box provides NAT and firewall. Phones work very well behind their box on a NATted address.
http://www.edgewaternetworks.com

by , Friday 04 of February, 2005 (21:35:25 UTC)
Article is not correct

NAT does not need to be avoided when doing SIP. With smart enough equipment and proper configuration it becomes trivial. Especially now when there are public STUN servers on the Internet.

There are 3 problems with doing SIP through NAT. Theres actually a 4th if you consider GW's trying to inherently do SPI (stateful packet inspection) on SIP, in the hopes of helping it do NAT traversal. The first problem is that device's try to Register with their private IP; or in point to point environment, putting their private IP as their VIA and Contact. The second problem with SIP through NAT is that GW/FW's are not going to allow inbound messages to your NATed device without an established pinhole (aka.. session). A session is created when an egress packet is sent from the NATed device to the Internet, the pinhole will allow the reply from the Internet to traverse the GW/FW and reach the NATed device. To keep a session up, which by the way are point to point, the NATed device behind the GW/FW must keep sending keep alive messages to keep the session/pinhole open. Sessions/pinholes have a life expectancy of about 30 secs to 5 minutes. So you wanna keep your keep alives at around 15-20 seconds... go with 10 and you should be good. The third problem was stated in the article, and thats the issues with RTP traffic traversing the NAT. I'll keep that one to myself as its a trade secret. ;)


by , Wednesday 22 of December, 2004 (19:16:05 UTC)
Incorrect Acronym for PAT

PAT equal Port Address Translation as opposed to Network Address Translation.

PAT=many addresses hiding behind one address
NAT= 1:1 ratio of private and real addresses

by , Tuesday 09 of November, 2004 (23:03:15 UTC)
Just port forward your listening and RTP ports

SIP and NAT isn't any bigger a deal than any other server application or p2p. Port forward the listening port and your range of RTP ports.

5060 and 10000 - 12000 (or however many RTP's you feel you need) # both UDP

In your sip.conf under general:

port = 5060 ; Port to bind to (probably don't change this)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine probably don't change this)
externip=xxx.xxx.xxx.xxx ; The IP that your ISP assigns you (change this)
localnet=192.168.1.0/255.255.255.0 ; your dhcp's local ip range and mask (maybe not required)
; (change to what your DHCP uses..the mask part probably doesn't need to change though)
srvlookup=yes ; resolves IP's? (maybe not required)
nat=yes ; required

by , Tuesday 02 of November, 2004 (01:04:13 UTC)
Broken link to Cisco White Paper

Cisco has (re)moved its White Paper "VOIP traversal of NAT and firewall". I have asked them to replace it or advise me of its new location. Meanwhile, it can be found here:
http://voip-itec.tamu.edu/files/reference/voip-nat.pdf


by , Saturday 09 of October, 2004 (02:14:37 UTC)
Re: Broken link to pdf

Hello,
The correct link is now http://www.sipcenter.com/sip.nsf/html/WEBB5YN5GE/$FILE/SIPNATtraversal.pdf

Kind regards,
Kathleen Misson
SIP Center Editor

by , Thursday 26 of August, 2004 (10:10:28 UTC)
Broken link to pdf

On page http://www.voip-info.org/wiki-NAT+and+VOIP the url under "NAT Traversal in SIP" which is http://www.sipcenter.com/files/SIPNATtraversal.pdf on 27 July 9pm CET gave error 404

by , Tuesday 27 of July, 2004 (19:07:17 UTC)