[Serusers] Help needed : SER / PSTN / NAT

olivier at siteboulevard.com olivier at siteboulevard.com
Wed Mar 17 18:13:32 CET 2004


Selon Klaus Darilion <klaus.mailinglists at pernau.at>:

many thanks.

In fact, everything work except no sound !

I put the debug on the rtpproxy and i see each time i call :

rtpproxy: command syntax error



I download the last src archive (not from cvs), and the last cvs version of 
rtpproxy.

What could it be ???


> 
> 
> olivier at siteboulevard.com wrote:
> > OK, thanks.
> > 
> > On the following conf (in fact, the NAT example), where should i put the 
> > rewritehost and forward function to my CISCO ??
> 
> In front of the lookup("alias") I would check if the username is 
> numerical, then I would format it (according to the local dial plan) to 
> an E.164 number. After that I would do an ENUM lookup.
> If after the ENUM lookup the request-URI is still an E.164 number, I 
> would rewrite the host.
> 
> Otherwise do the lookup-alias and lookup location.
> 
> In but cases, the message will be forwarded by the t_relay at the end of 
> your script.
> 
> Klaus
> 
> 
> 
> 
> 
> > 
> > #
> > # $Id: nathelper.cfg,v 1.1.2.1 2003/11/24 14:47:18 janakj Exp $
> > #
> > # simple quick-start config script including nathelper support
> > 
> > # This default script includes nathelper support. To make it work
> > # you will also have to install Maxim's RTP proxy. The proxy is enforced
> > # if one of the parties is behind a NAT.
> > #
> > # If you have an endpoing in the public internet which is known to
> > # support symmetric RTP (Cisco PSTN gateway or voicemail, for example),
> > # then you don't have to force RTP proxy. If you don't want to enforce
> > # RTP proxy for some destinations than simply use t_relay() instead of
> > # route(1)
> > #
> > # Sections marked with !! Nathelper contain modifications for nathelper
> > #
> > # NOTE !! This config is EXPERIMENTAL !
> > #
> > # ----------- global configuration parameters ------------------------
> > 
> > debug=7         # debug level (cmd line: -dddddddddd)
> > fork=yes
> > log_stderror=yes        # (cmd line: -E)
> > 
> > /* Uncomment these lines to enter debugging mode 
> > 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"
> > 
> > # ------------------ module loading ----------------------------------
> > 
> > # Uncomment this if you want to use SQL database
> > #loadmodule "/usr/local/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/textops.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"
> > 
> > # !! Nathelper
> > loadmodule "/usr/lib/ser/modules/nathelper.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 interval 30 s
> > #modparam("nathelper", "ping_nated_only", 1)   # Ping only clients 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;
> >         };
> > 
> >         # !! Nathelper
> >         # Special handling for NATed clients; first, NAT test is
> >         # executed: it looks for via!=received and RFC1918 addresses
> >         # in Contact (may fail if line-folding is used); also,
> >         # the received test should, if completed, should check all
> >         # vias for rpesence of received
> >         #if (nat_uac_test("3")) {
> >                 # Allow RR-ed requests, as these may indicate that
> >                 # a NAT-enabled proxy takes care of it; unless it is
> >                 # a REGISTER
> > 
> >                 if (method == "REGISTER" || ! search("^Record-Route:")) {
> >                     log("LOG: Someone trying to register from private IP, 
> > rewriting\n");
> > 
> >                     # This will work only for user agents that support
> symmetric
> >                     # communication. We tested quite many of them and
> majority 
> > is
> >                     # smart enough to be symmetric. In some phones it takes
> a 
> > configuration
> >                     # option. With Cisco 7960, it is called NAT_Enable=Yes,
> 
> > with kphone it is
> >                     # called "symmetric media" and "symmetric signalling".
> > 
> >                     fix_nated_contact(); # Rewrite contact with source IP
> of 
> > signalling
> >                     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
> >                 };
> >         #};
> > 
> >         # 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();        
> > 
> >         # subsequent messages withing a dialog should take the
> >         # path determined by record-routing
> >         if (loose_route()) {
> >                 # mark routing logic in request
> >                 append_hf("P-hint: rr-enforced\r\n"); 
> >                 route(1);
> >                 break;
> >         };
> > 
> >         if (!uri==myself) {
> >                 # mark routing logic in request
> >                 append_hf("P-hint: outbound\r\n"); 
> >                 route(1);
> >                 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("iptel.org", "subscriber")) {
> > #                                www_challenge("iptel.org", "0");
> > #                                break;
> > #                        };
> > 
> >                         save("location");
> >                         break;
> >                 };
> > 
> >                 lookup("aliases");
> >                 if (!uri==myself) {
> >                         append_hf("P-hint: outbound alias\r\n"); 
> >                         route(1);
> >                         break;
> >                 };
> > 
> >                 # native SIP destinations are handled using our USRLOC DB
> >                 if (!lookup("location")) {
> >                         sl_send_reply("404", "Not Found");
> >                         break;
> >                 };
> >         };
> >         append_hf("P-hint: usrloc applied\r\n"); 
> >         route(1);
> > }
> > 
> > route[1] 
> > {
> >         # !! Nathelper
> >         if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&
> !
> > search("^Route:")){
> >             sl_send_reply("479", "We don't forward to private IP
> addresses");
> >             break;
> >         };
> >         
> >         # if client or server know to be behind a NAT, enable relay
> >         if (isflagset(6)) {
> >             force_rtp_proxy();
> >         };
> > 
> >         # NAT processing of replies; apply to all transactions (for
> example,
> >         # re-INVITEs 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]") {
> >         fix_nated_contact();
> >         force_rtp_proxy();
> >     # otherwise, is it a transaction behind a NAT and we did not
> >     # know at time of request processing ? (RFC1918 contacts)
> >     } else {
> >         fix_nated_contact();
> >     };
> > }
> > 
> > 
> > 
> > 
> >>
> >>olivier at siteboulevard.com wrote:
> >>
> >>>Hi,
> >>>
> >>>After some testing on the latest release, i have some problem doing the 
> >>>following on LINUX :
> >>>
> >>
> >>latest? du you mean unstable or latest stable?
> >>
> >>
> >>>Scenario :
> >>>- SIP Phones behind a NAT
> >>>- SER server under linux with rtpproxy launched
> >>>- a 3660 cisco gateway with PSTN connectivity enabled.
> >>>
> >>>When i call with SIP phone a PSTN number, everything is OK BUT no sound 
> >>>anywhere.
> >>>
> >>
> >>Use ethereal to verfiy that the SDP in the INVITE and 200 OK (or 183 
> >>Early Media) are rewritten by nathelper&rtpproxy to point to the IP:port 
> >>of the rtpproxy. If this is correct, you should see RTP streams to 
> >>rtpproxy (which should be forwarded to the GW and the NAT box)
> >>
> >>
> >>>I could not find a sample ser.cfg script that reflect this scenario.
> Could
> >>
> >>>someone send me this scenario ?
> >>
> >>this is like any other scenario with a client behind NAT and one client 
> >>with public IP.
> >>
> >>
> >>>Maybe i missunderstood some things. In particular, do i need to launch
> two
> >>
> >>>instances of ser (one for outbound proxy, another for request. If yes,
> how
> >>
> >>to 
> >>
> >>>do that)
> >>
> >>You don't need two instances.
> >>
> >>Klaus
> >>
> >>
> > 
> > 
> > 
> > _______________________________________________
> > Serusers mailing list
> > serusers at lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> > 
> > 
> 
> 





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