Unless you have a router capable of handling SIP, you need to better understand SIP. In handling SIP NAT traversal with a router not capable of handling SIP, you will find the audio or call signalling (or both) will fail.
SIP is not a simple NAT, the RTP ports ranges and such do not matter, because SIP is a peer-topeer UDP (typically) media stream. This media stream is packetized, with headers with the source/destination in it by DNS name or IP address. So while a call could be established, since it goes through a proxy, the media (or sound) will fail. NAT for SIP also has three major components, so in essence, a specialized approach is needed that will take into account the header, RTP and related parts of the transmission. In you case, if you are simply trying to"call home" from a softphone to a sipx installation behind NAT, this can be done with an IPSEC VPN connection.. It works very well if your VPN pushes your local domain out on connection, so your softphone config stays the same. Perhaps you can share what your intent is so others can chime in and help you understand fully what is needed? Thanks, Tony >>> Roger Toennis <[EMAIL PROTECTED]> 07/13/08 17:58 PM >>> <snip> I have godaddy set to direct sipx.<my domain>.com to my external IP on my router and my router is port forwarding 8443 so I can manage the sipx remotely. I guess maybe what I need is to set up an SRV rules on godaddy for the domain for the SIP traffic so when the X Lite phones send sip traffic to "sipx.mydomain.com" it gets forwarded to my router IP. Router then needs to port forward that traffic on the known ports on sipx?? Does that make sense? And if so what are the specifics I need to setup for port forwarding 5060, etca al. TCP? UDP? both? RTP media from external? what port ranges? ??? Thanks, Roger <snip> On Jul 13, 2008, at 3:09 PM, Scott Lawrence wrote: > > On Sun, 2008-07-13 at 14:33 -0600, Roger Toennis wrote: >> Centos 3.10.1 single disk install >> on home LAN behind firewall >> No local DNS on SIPX...... >> >> So I was able to get 2 users configured and register those users with >> XLite softphones from PCs on my local LAN. >> However when I try making calls between the registered extension the >> calls failed with "user unavailable". >> >> What am I missing that I havn't done? > > http://sipx-wiki.calivia.com/index.php/ > SipX_Phone_Registration_Troubleshooting > > -- > Scott Lawrence tel:+1.781.229.0533;ext=162 or > sip:[EMAIL PROTECTED] > sipXecs project coordinator - SIPfoundry http:// > www.sipfoundry.org/sipXecs > CTO, Voice Solutions - Bluesocket Inc. http://www.bluesocket.com/ > http://www.pingtel.com/ > > -- Roger Toennis President/CEO Liquid Media Software LLC 303-522-5477 "Making the simple complicated is commonplace. Making the complicated simple, that's creativity" - Charles Mingus CONFIDENTIALITY NOTICE: This e-mail, and any attachments thereto, is intended only for use by the addressee(s) named herein and may contain privileged and/or confidential information. If you are not the intended recipient of this e-mail, you are hereby notified that any dissemination, distribution or copying of this e-mail, and any attachments thereto, is strictly prohibited. _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
