Re: [Users] Mediaproxy newbie

Hello,

maybe this link helps:

http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy

Cheers, Daniel

On 10/18/06 19:04, Marnus van Niekerk wrote:

Hi,

I am trying to set up openser with mediaproxy (at xx.xx.xx.133) to route calls from UA behind NAT to asterisk as voicemail (at xx.xx.xx.134) and PSTN gateways (at xx.xx.xx.32)

I can see in the SDP payload that the RTP is being sent from asterisk to mediaproxy, but in sessions.py it shows the private ip not the public one and I have one way audio.

Can anybody help please.

opnser.cfg below.

Marnus

--

debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL6

check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo"

# ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so"

# Logging loadmodule "/usr/local/lib/openser/modules/xlog.so"

loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so"

# ----------------- setting module-specific parameters ---------------

# -- usrloc params -- modparam("usrloc", "db_mode", 0)

modparam("usrloc", "db_mode", 2)

modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password")

modparam("rr", "enable_full_lr", 1)

#tm timeout for voicemail params modparam("tm", "fr_timer", 3) modparam("tm", "fr_inv_timer", 35) modparam("tm", "noisy_ctimer", 1)

# parms for NAT/mediaproxy modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy", "natping_interval", 30) modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "sip_asymmetrics", "/usr/local/etc/openser/sip-clients") modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/openser/rtp-clients") modparam("registrar", "nat_flag", 6)

# ------------------------- request routing logic -------------------

# main routing logic route { #check for old messages: could mean a problem with the DNS entries or some other loop-causer... if (!mf_process_maxfwd_header("10")) { xlog("L_WARN", "WARNING: Too many hops\n"); sl_send_reply("483", "Too many hops, forward count exceeded limit\n"); return; };

   #check for extremely large messages; we don't need a sip dos 

attack if (msg:len >= 2048) { xlog("L_WARN", "WARNING: Message too large, &>= 2048 bytes\n"); sl_send_reply("513", "Message too large, exceeded limit\n"); return; };

   # Track what is happening
   xlog("L_INFO", "SIP Request: method [$rm] from [$fu] to [$tu]\n");

   #record everything besides registers and acks
   if(method!="REGISTER" && method!="ACK")
   {
           setflag(1);
   };

   # Record Route Section
   if (method=="INVITE" && client_nat_test("3"))
   {
           record_route_preset("xx.xx.xx.133:5060;nat=yes");
   }
   else if (method!="REGISTER")
   {
           record_route();
   }

   # Call tear down section
   if (method=="BYE" || method=="CANCEL")
   {
           end_media_session();
   }

   #do not send to voicemail if BYE or CANCEL
   #is used to end call before user pickup or timeout
   if(method=="CANCEL" || method=="BYE")
   {
           setflag(10);
   };

   #grant route if route headers already present
   if (loose_route())
   {
           # May need client_nat_test & use_media_proxy here...
           route(1);
           return;
   };

   #Always require authentication, which could result in a PSTN
   if (method=="REGISTER")
   {

           if (!search("^Contact:[ ]*\*") && client_nat_test("7"))
           {
                   setflag(6);
                   fix_nated_register();
                   force_rport();
           };

           if(!www_authorize("domain.tld", "subscriber"))
           {
                   www_challenge("domain.tld", "0");
                   return;
           }
           else
           {
                   if (!check_to())
                   {
                           sl_send_reply("401", "Unauthorized");
                           return;
                   };

                   #Save into user database, used below when 

checking if user is available xlog("L_INFO", "REGISTER: User $fu Authenticated Correctly\n"); save("location"); return; }; };

   if (method=="INVITE")
   {
           if (client_nat_test("3"))
           {
                   setflag(7);
                   force_rport();
                   fix_nated_contact();
           };

           if(uri=~"sip:\*86@.*")
           {
                   #authorize if a call is going to VM
                   if(!proxy_authorize("domain.tld", "subscriber"))
                   {
                           proxy_challenge("domain.tld", "0");
                           return;
                   };

                   xlog("L_INFO", "CALL: Call from $fu to check 

voicemail\n"); rewritehostport("vm.domain.tld:5060"); } else { if (does_uri_exist()) { #Call is to sip client, so do nothing but route xlog("L_INFO", "CALL: Sip client\n"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); xlog("L_ERROR", "ERROR: User $tu Not Found\n"); return; }; } else { #authorize if a call is going to PSTN if(!proxy_authorize("domain.tld", "subscriber")) { proxy_challenge("domain.tld", "0"); return; };

                           #Call destination is PSTN, so send it 

to the gateway xlog("L_INFO", "CALL: PSTN $tu from $fu \n"); rewritehostport("ast1.domain.tld:5060"); }; };

           #Make sure that all subsequent requests go through us;
           #done at the top already
           #record_route();
   }
   else
   {
           if (does_uri_exist())
           {
                   #Call is to sip client, so do nothing but route
                   xlog("L_INFO", "CALL: Sip client\n");
                   if (!lookup("location"))
                   {
                           sl_send_reply("404", "Not Found");
                           xlog("L_ERROR", "ERROR: User $tu Not 

Found\n"); return; }; } else { #Call destination is PSTN, so send it to the gateway xlog("L_INFO", "CALL: PSTN $tu from $fu \n"); rewritehostport("ast1.domain.tld:5060"); }; #record_route(); };

   #ALL PROCESSING IS DONE, SO ROUTE
   route(4);
   route(1);

}

route[1] { #send the call outward if(method=="INVITE" && !isflagset(10)) { t_on_failure("2"); # voicemail if failure };

   if (!t_relay())
   {
           xlog("L_WARN", "ERROR: t_relay failed");
           sl_reply_error();
   };

}

# ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- route[4] { if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; }

failure_route[2] { if(!t_was_cancelled() && !t_check_status("407")) { revert_uri(); rewritehostport("vm.domain.tld:5060"); append_branch(); #PREVENT SOME CRAZY VOICEMAIL LOOP xlog("L_INFO", "INFO: CALL TO VOICEMAIL"); setflag(10); route(1); } }

onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; };

   if (client_nat_test("1"))
   {
           fix_nated_contact();
   };

}