Re: [Serusers] Help needed : SER / PSTN / NAT

Selon Klaus Darilion <klaus.mailinglists(a)pernau.at>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(a)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

> >># 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

> >a > >>configuration >> # option. With Cisco 7960, it is called

> >>with kphone it is >> # called "symmetric media" and "symmetric

>> >> 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(a)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

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