Hi all,
I would like to thank everyone for their help to date answering my questions (particularly Giovanni). The clients now register with SER despite the NAT issues. I have yet to test a call so I may be back with more queries if the voice doesnt transmit!!
I used the default NAT script provided on the voip inof wiki. However I want my clients to authenticate before they are allowed to register. If I enable authentication in the ser.cfg script, registration fails. I do not understand why as the mysql database is set up and I have created user accounts.
I have included my ser.cfg script below with the authentication part and the mysql load module part commented out. Users only authenticate at the moment if these parts are commented. Any help would be greatly appreciated.
Also on a slightly unrelated note, I am trying to test this system with Grandstream Budgetone 100 hardphones (Its so far tested with XLite)I plug these hones into a hub which is in turn plugged into a router,however the phone wont obtain an IP address through dhcp or take its statically assigned one. Shouldn't an ip phone act like any other IP device on a network i.e. a pc etc??
Thanks as always, Aisling.
# # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script #
# ----------- global configuration parameters ------------------------
#debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */
check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 #children=4 fifo="/tmp/ser_fifo"
alias=84.203.148.14
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database #loadmodule "/usr/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/nathelper.so" #loadmodule "/usr/lib/ser/modules/mediaproxy.so" loadmodule "/usr/lib/ser/modules/textops.so" #loadmodule "/usr/lib/ser/modules/maxfwd.so"
# Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2)
# -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password")
# -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1)
#!!Nathelper #modparam("registrar","nat_flag",6) #modparam("nathelper","natping_interval",30) #Ping intervals 30 seconds #modparam("nathelper","ping_nated_only",1) #Ping only clinets behind NAT
# -------------------------request routing logic-------------------
# main routing logic
route{
# initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; };
#############Aisling Insert################ # #!Nat Insert # #the below line tests if the IP of the received packet is different from the IP in the via header and also # #sees if the IP address in the contact header is private # if (nat_uac_test("3")){ # if (method == "REGISTER" || ! search("^Record-Route:")){ # log("Log: Someone trying to register from private IP,rewriting\n"); # # fixed_nated_contact(); #Rewrite contact with source IP # if (method == "INVITE"){ # fix_nated_sdp("1"); #Add direction=active to SDP # }; # force_rport(); # Add rport parameter to topmost Via # setflag(6); # Mark as Nated # }; # }; ###################End#####################
# we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol
if (!method == "REGISTER") record_route();
# loose-route processing if (loose_route()) { t_relay(); break; };
# if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication # if (!www_authorize("84.203.148.14", "subscriber")) { # www_challenge("84.203.148.14", "0"); # break; # }; save("location"); break; };
# native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; };
#inserted by klaus if (method == "INVITE"){ record_route(); force_rtp_proxy(); /* set up reply processing*/ t_on_reply("1"); };
# forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); };
}
#insert by klaus
onreply_route[1]{ if (status=~"[12][0-9][0-9]") force_rtp_proxy(); }
#route[1] #{ #if client or server know to be behind NAT, enable relay # if (isflagset(6)){ # force_rtp_proxy(); # }; # # #NAT processing of replies; apply to all transaction (for example, # #reinvites from public to private UA are hard to identify as # #Nated at the moment of request processing); look at replies # t_on_reply("1"); # # #send it out now; use stateful forwarding as it works reliably # #even for UDP2TCP # if(!t_relay()){ # sl_reply_error(); # }; #}
#!!NatHelper
#onreply_route[1]{ #Nated Transaction?? #if (isflagset(6) && status =~ "(183)\2[0-9][0-9]"){ # #fixed_nated_contact(); # force_rtp_proxy(); # } #else if (nat_uac_test("1")){ # fix_nated_contact(); # }; #}
---- Original Message ---- From: jev@emmplus.ie To: ashling.odriscoll@cit.ie Subject: Re: [Serusers] FW: SER and NAT Date: Fri, 21 Jan 2005 13:47:47 -0800
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Hi
I have installed ser with STUN support, since my routers are not symmetric, in theory this should solve natting issues, however I seem to get completely random results. I am using xlite softphones behind the SAME nat to make a call, they can call out to pstn (via asterisk), however one phone can call the other across the internal network, but not vice versa, even though both PC's are on the same network. The ser/asterisk/stun boxes are on apublic network.
I came across a post which using a tcpdump would draw a diagram of the sip call path, does anyone have the url/name for that, I just cant see what wrong in the sip debug messages, maybe a pic will help me out.
tks
Iqbal
-
Ashling O'Driscoll -
Iqbal Gandham