Hi I am still trying to solve this issue: ...using avpops to enable SER to see if the UAs are located behind the same address and then trying to let the UAs speak directly to each other (without force rtpproxy), please see my config. below and send some advice for what do I have to do in route 3? ... if (!lookup("location")) { log(1,"unable to locate user X ... sending to route(4)! \n"); # handle user which was not found route(4); break; }; ### Test if UAS are in the same NAT: # get the host part of the final uri (IP part) and store it in AVP ID 13 avp_write("$ruri/domain", "i:13"); if (avp_check("i:13","eq/$src_ip/i")) { log(1, "source IP is the same as destination IP\n"); route(3) ;t_on_reply("3"); break; }; avp_delete("i:13/g"); ... # -------------- Same NAT Call Routing (no force rtpproxy) ---- route[3]{ log(1," route[3]: UAs are in the same nat, NO force_rtp_proxy "); fix_nated_contact(); if(!t_relay()){ sl_reply_error(); }; log(1, "Route[3]: Send it out now!!!\n"); } thanks rafael

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I have looked into the problem related to saving saving resources by not calling rtpproxy or mediaproxy when both the caller and callee are behind the same NAT. This topic has been discussed many times however, there have been no "working" examples, or even partial examples posted (that I'm aware of). Below is a solution to the problem, but is a BAD hack at best. This solution is BAD because AVP's are added to the usr_preferences table during the registration process, but there is no notification that a registration has expired and is no longer valid. A cron job could be run periodically to delete all received_ip AVP's with no matching entry in the location table, but this is another hack to fix the first hack. The solution is BAD because a database calls must be made to save the received_ip address into an AVP, then another database call must be made to reload the value just so that it can be tested against the caller's $src_ip. A pseudo variable $Ri already exists that contains the value we use in received_ip, but I have not been able to find a way for avp_check() to use $Ri. The solution is BAD because it will fail when the SIP device(s) are behind more than a single NAT router. I believe this is not an issue for 95% of the users and 99% of the small office or home users. I think that a better solution would be to enhance avp_check() to accept pseudo variables for the "value" parameter. This would remove the need for any database calls. I'm cross-posting this response to the developers list to bring this to their attention and ask for suggestions (it's possible that I've totally missed something). If a decision is made to open the avpops module to pseudo variables (other than just avp_print()), I would suggest the following be looked at: avp_check() the value parm should accept pseudo variables avp_write() the value parm should accept pseudo variables avp_pushto() the name parameter should accept pseudo variables avp_op() the value parameter should accept pseudo variables Regards, Norm 1) During REGISTER processing, place the following code. I would suggest that you put the code "after" all authentication checks and "after" the save("location") statement. # Delete any previously saved IP addressess from the user. avp_db_delete("$from/username","s:received_ip"); # Save the source IP address of the user into an AVP called received_ip. # The saved IP address should be the public address of the NAT router. avp_write("$src_ip", "s:received_ip"); # Save the AVP received_ip into the usr_preferences tables associated with the user that just registered avp_db_store("$from/username","s:received_ip"); 2) During INVITE processing (or wherever you make the decision to use or not use rtpproxy / mediaproxy) , place the following code. #--------------------------------------------------------------------------- # Are the caller and callee behind the same NAT ? #--------------------------------------------------------------------------- avp_db_load("$to/username", "s:received_ip"); if (avp_check("s:received_ip", "eq/$src_ip")) { setflag(CALLER_AND_CALLEE_ ARE_BEHIND_THE_SAME_NAT); }; _______________________________________________ Devel mailing list Devel(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/devel

could be an issue of the local gateway. most NAT devices explicitly block "hairpinning", which is happening in your case. -h Jan Henkins wrote: >Hello Rafael, > >This is an issue I'm having at the moment as well, where two UA's are >behind the same nat. SIP obviously works fine, I can ring one UA from >the other (one UA a Grandstream Budgetone 100 and the other the X-Lite >softphone from Counterpath). Audio doesn't work. Did you get this >working with forcing rtpproxy? > >On Fri, 2005-12-02 at 09:44 -0500, Rafael R. GV wrote: > > >>Hi >>I need to let the UAs speak directly to each other without force >>rtpproxy, Is there any solution for this issue? >> >>rafael >> >> >>---------- Forwarded message ---------- >>From: Rafael J. Risco G.V. <rafael.risco(a)gmail.com> >>Date: Apr 22, 2005 2:17 PM >>Subject: UA´s behind same NAT using na thelper/rtpproxy, No Audio... >>To: serusers(a)lists.iptel.org >> >>I am using avpops to enable SER to see if the UAs are located behind >>the same address and now I need to let the UAs speak directly to each >>other (without force rtpproxy), please someone see my configuration >>below and send some advice. >> >>rafael >> >>----. >> if (!lookup("location")) { >> log(1,"unable to locate user \n"); >> # handle user which was not found >> route(4); >> break; >> }; >> >> ### Test if UAS are in the same NAT: >> # get the host part of the final uri (IP part) >> # and store it in AVP ID 13 >> >> avp_write("$ruri/domain", "i:13"); >> if (avp_check("i:13","eq/$src_ip/i")) { >> log(1, "source IP is the same as destination IP >>\n"); >> route(3); >> break; >> }; >> avp_delete("i:13/g"); >>. >>. >># -------------- Same NAT Call Routing (no force rtpproxy) ---- >> >>route[3]{ >> log(1," route[3]: UAs are in the same nat, NO force_rtp_proxy >>"); >> >>### Do something!!!! >> >> if(!t_relay()){ >> sl_reply_error(); >> }; >>} >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >>

If the gateway has no issues with hairpinning, then yes, it's quite possible. I use a Linksys gateway at home and a slightly older, pre-sip-proxy version of Astaro linux firewall at work, and we have multiple UAs behind each in the NAT space of our firewall. They can call each other. They can all outside. All based off registrations with a SER server on the outside of the network.

Netgear specifically has some serious issues both with hairpinning and with just plain ol' SIP. Netgear makes some mighty unfriendly gateways. :)

If you can't do it, though, it makes sense to set up some sort of proxy on the inside of the NAT that all the UAs register with, and have it pass things back and forth... forwarding the necessary data from outside to the server on the inside using port-fowarding rules. For some of our customers, we've recommended Asterisk setups inside their NAT, just to make the passing of RTP packets more rational. You don't have to worry about individual client UA RTP settings, you can just worry about forwarding the RTP ports to Asterisk, and then inside the NAT do anything you wish. Since SER doesn't manage RTP, using just SER becomes problematic if your UAs are not homogeneous.