[Serusers] Re: NAT & SER
szj
zjsun at biigroup.com
Fri Mar 4 01:55:22 CET 2005
Ozan Blotter wrote:
> Dear Sun Zongjun,
>
> I've read and can you please check this for me >>>
>
>
> Conditions as follows :
>
> * SER runs on a Public IP
> * SER works without auth & database modules,
> * Nearly all user behind NAT (but routers configured to do port
> forwarding for TCP/UDP 5060) to help SER in some cases,
> * Users numbers in format of 833XXXXXXX 834XXXXXXX and they should
> call each P2P-SIP-Calls (if not behind NAT),
> * If a user need to call PSTN end point (SIP Gateway located at
> 212.154.32.154) the call traffic should flow over SER to SIP Gateway
> via T1 connection already located between that systems so SER handles
> all voice traffic by help of RTP Proxy.
> * UA's registers on SER (Zyxel Prestige 2000, Zyxel Prestige 200W,
> Cisco ATA186 etc.)
>
> Problem is users cannot call each other (if i comment lines for
> nathelper they can call)
>
> It's clear i think, and below is my ser.cfg, what do i need extra or
> erase.
Sorry, I feel shame that I can't see anything wrong about it. maybe you can
add the following line:
rtpproxy_sock="unix:/foo/bar=4 udp:1.2.3.4:3456=3 udp:5.6.7.8:5432=1"
I think the seruser mail list may have the same questions. You can
consult them.
BTW I use the iptel.org's SER to test my SIP UA, Not my own SER with
nathelper.
Best Regards.
Sun Zongjun
>
>
> <-<-<-<-< MY SER.CFG STARTS HERE >->->->->
>
> #
> # $Id nathelper.cfg,v 1.1.2.1 20050301 by Ozan Blotter 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=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"
>
> # ------------------ module loading ----------------------------------
>
> # Uncomment this if you want to use SQL database
> loadmodule "/usr/local/lib/ser/modules/sl.so"
> loadmodule "/usr/local/lib/ser/modules/tm.so"
> loadmodule "/usr/local/lib/ser/modules/rr.so"
> loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
> loadmodule "/usr/local/lib/ser/modules/usrloc.so"
> loadmodule "/usr/local/lib/ser/modules/registrar.so"
> loadmodule "/usr/local/lib/ser/modules/textops.so"
>
> # !! Nathelper
> loadmodule "/usr/local/lib/ser/modules/nathelper.so"
>
> # ----------------- setting module-specific parameters ---------------
>
> # -- usrloc params --
>
> modparam("usrloc", "db_mode", 0)
>
> # -- 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", 10) # Ping interval 10 seconds
> 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;
> };
>
> # if ((method=="NOTIFY")&& search("^Event: Keep-Alive")) {
> # ls_send_reply("200","OK");
> # 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\$
>
> # 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 $
> # 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") {
>
> if (!(uri=~"sip:(833)|(834)")) {
> t_relay_to_udp("212.154.32.154","5060");
>
> 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])\.)"
> && !searc$
> 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();
>
> if (!search("^Content-Length:\0")){
> force_rtp_proxy();
> };
> # otherwise, is it a transaction behind a NAT and we did not
> # know at time of request processing (RFC1918 contacts)
> } else if (nat_uac_test("1")) {
> fix_nated_contact();
>
> if (!search("^Content-Length:\0")){
> force_rtp_proxy();
> };
> # otherwise, is it a transaction behind a NAT and we did not
> # know at time of request processing (RFC1918 contacts)
> } else if (nat_uac_test("1")) {
> fix_nated_contact();
> };
> }
>
> <-<-<-<-< MY SER.CFG ENDS HERE >->->->->
>
>
> > szj wrote:
> >
> > During the testing procedure, I found that when both sip UAs
> > who are located behind the same NAT cloud want to establish
> > voice or video connection, there is not neccesary to bridge
> > them with rtpproxy. Only in situations where one of UA sits
> > behind NAT or each UA sits behind different NAT clouds, that
> > need a RTPProxy to bridge their media stream.
> > What I mean is SER can determine the use of RTPproxy or not
> > through the registration of sip UA. In location table, there
> > are recieved and contact fields. But ser don't fill the
> > recieved field, I think it is very userful for NAT.
> >
> > Glad to hear your instructions
> > Best Regards
> >
> > Sun Zongjun
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