Derek, You said - Needless to say when I don't have any NAT settings on the SIP phone I don't get any registration with the * server (this confuses me too - I'm
not sure why I only get registration when I set the * server to be the outbound proxy? Maybe its because the SIP phone sends its local IP in the RTP packets?). SIP is not NAT friendly (unlike IAX) and yes your device will try to send its local IP (in SIP packets), unless in the case of a budgetone phone you set the 'Use NAT IP' to your external IP addr. You will also have to NAT the public ip for the SIP port (5060?) and RTP ports (whatever) to your phones private IP. Must admit not tried it myself, but happy to jointly experiment if you like? _______________________________________________ Ray _______________________________________________ -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Derek Conniffe Sent: 13 September 2005 12:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Nat & Sip & Pain Hi everyone, I decided to have a look at SIP & NAT again and I've been at it for a [quite a] few hours but typically nothing is working for me. Actually I'm not sure if SIP and NAT can ever work but some emails on this list do suggest that someone has got it working, once, maybe. I'm experimenting with a ZyXEL 2000W [WiFi Sip phone] which supports "Outbound Proxy", "STUN" and "Fake WAN Address on SIP and RTP". I'm using Netfilter (IPTables) on Linux as the Firewall at NAT gateway to the Internet. I'm lacking knowledge in UDP, RTP and SIP - which doesn't help of course. In my experiments the only thing that seems to allow me to make a call is to enter the [public Internet] IP address of my * server into the "Outbound Proxy" setting in the SIP phone - then it registers and I can make a call but no audio, either direction, is heard. I would have thought that the "Outbound Proxy" should be inside the NAT gateway but then I read the settings for a Budgetone BEHIND nat on the FWD webpage (http://www.freeworlddialup.com/support/configuration_guide/configure_yo ur_fwd_certified_phone/grandstream_budgetone/outbound_proxy) where they suggest that the Outbound Proxy should be an external Internet public proxy server ? Then I was reading about STUN and what a nice sounding solution it is - so I downloaded and installed the Vivida STUN server - compilation & installation was nice and easy and I set the STUN primary IP address & port into the SIP phones STUN servers settings. I could see that the SIP phone communicated with the STUN server (lots of stuff about mapping between my local NAT gateway's public IP address and the secondary IP address of the STUN server)... but no registration or [apparent] communication with the * server. I didn't try to do anything with the "Fake WAN address.." settings or try to redirect incoming UDP ports from the firewall to the SIP phone because I'm trying to see if its possible to setup a deploy-anywhere SIP phone solution. Needless to say when I don't have any NAT settings on the SIP phone I don't get any registration with the * server (this confuses me too - I'm not sure why I only get registration when I set the * server to be the outbound proxy? Maybe its because the SIP phone sends its local IP in the RTP packets?). Does anyone know how to get NAT & SIP working where the SIP phone is behind a NAT server talking to a publicly accessible * server? Thanks for any help! When I run FWD's "netcheck" on my local PC (also behind the NAT) I get: Internet Connection: Connected, Direct/NAT: Using NAT, NAT type: Port Restricted Nat, NAT UPnP enabled: No, Local IP Address: 192.168.5.10, WAN IP Address: XXX.XXX.XXX.XXX (public IP address), Port 5060: Blocked, port 5082: Blocked. [Maybe] useful Links that I've found on my Nat & SIP travels:- http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions ------------------------------------------------------------- Here VOIP INFO claim that "Asterisk as a SIP server outside nat, clients on the inside connecting to Asterisk" is "solved" with "with nat <tiki-index.php?page=Asterisk+sip+nat>=yes and qualify <tiki-index.php?page=Asterisk+sip+qualify>=xxx in sip.conf <tiki-index.php?page=Asterisk+config+sip.conf> for the client in most cases. Some clients (X-lite) assist themselves by using STUN <tiki-index.php?page=STUN> and sending UDP keep-alive packets. Qualify <tiki-index.php?page=Asterisk+sip+qualify> sends keep-alive packets from Asterisk to the client on the inside." - however I can't get it to work http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asteris k.html ------------------------------------------------------------------------ ----------- Here there is some detail about the NAT= option in sip.conf and firewall NAT types plus some understandable diagrams of why SIP & NAT is so much bother. http://www.voip-info.org/wiki-STUN -------------------------------------- The VOIP INFO page about STUN - I don't think I learned much here - except the link to the Vovida STUN server software Asterisk Users - Email from [EMAIL PROTECTED] - 02/July/2005 23:49 -------------------------------------------------------------------- Thierry claims that you need to put special MASQUERADE POSTROUTING rules into iptables to make it NAT UDP properly - tried it but didn't work for me Asterisk Users - Email from [EMAIL PROTECTED] - 16/Aug/2005 10:29 ------------------------------------------------------------------------ Kamran Ahmad sounds like someone who [might have] had SIP & NAT working - until it wasn't working.... BTW My Current SIP sip.conf entry that I'm using for testing (which doesn't work of course!): - [0035314401789] context=PublicSip type=friend port=5060 username=0035314401789 password=XXXXXXXX callerId=0035314401789 nat=route ; assume a NAT connection (note: route doesn't seem to make any difference compared to "yes") qualify=yes ; keep-alive packets to keep NAT SIP open insecure=yes ; insecure and auth don't seem to make things work any better/worse! auth=plaintext ; host=dynamic ; and with a dynamic IP address canreinvite=no ; always keep asterisk in the media path ;dtmfmode=info ; could be inband ? dtmfmode=rfc2833 ; could be inband ? but doesn't matter - still NAT & SIP isn't working [EMAIL PROTECTED] disallow=all ;allow=ilbc ;allow=ulaw allow=g729 ;allow=ulaw ;allow=all -- Derek Conniffe Rivertower Ltd Ireland: (Freephone) 1800 719 400 Ireland: (Local) 01 244 9719 United Kingdom: 0870 068 2368 International: 00 353 1 244 9719 Derek Conniffe DDI: 01 201 0146 (International: 00 353 1 201 0146) Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 856 3823) Fax: 01 201 0085 (International: 00 353 1 201 0085) Email: [EMAIL PROTECTED] Web: http://www.rivertowerhosting.com _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users