Hi Friends,
I am trying to learn ser and wrap my head around the routing logic. My first project is a simple outbound proxy to handle SIP/RTP from an SPA2000 behind a NAT. The ser server is on the public Internet, but I am having trouble making it work :( The spa2000 has been reset to default, and basically setup the same way that I'd set it up for FWD behind nat).
I am using an example config from the Internet as as starting point (included below) and running on Ser 0.8.14 on FreeBSD (from a port). I am using the RTP proxy from ser cvs. This , It almost works, and capturing the traffic with ethereal looks to be mostly correct, but I am seeing duplicate sip messages (plz excuse formatting), which I think is causing me a big problem (even if it isn't, I'd like to know why this is happening).
I didn't include frames 1 and 2, they are a SIP keepalive.
3 is the request from the spa2000 to the proxy 4 is the response from the proxy 5 is the request from the proxy to the itsp 6 is a dup! And then the problem compounds as the itsp tries to connect the same call twice.
I would be very appreciative of any advice from you veterans ;-)
Thanks, Tim
** Doctored trace
- 207.46.199.15 is the address of the NAT - 207.46.199.14 is the address of the Proxy running ser - 69.16.138.164 is the address of my itsp's SIP proxy
3 4.563455 207.46.199.15 207.46.199.14 SIP/SDP Request: INVITE sip:6415551234@my.itsp.com, with session description 4 4.564809 207.46.199.14 207.46.199.15 SIP Status: 100 trying -- your call is important to us 5 4.566539 207.46.199.14 69.16.138.164 SIP/SDP Request: INVITE sip:6415551234@my.itsp.com, with session description 6 4.578979 207.46.199.14 69.16.138.164 SIP/SDP Request: INVITE sip:6415551234@my.itsp.com, with session description 7 4.589856 69.16.138.164 207.46.199.14 SIP Status: 100 Trying 8 4.602580 69.16.138.164 207.46.199.14 SIP Status: 407 Proxy Authorization Required 9 4.602733 207.46.199.14 69.16.138.164 SIP Request: ACK sip:6415551234@my.itsp.com 10 4.602808 207.46.199.14 207.46.199.15 SIP Status: 407 Proxy Authorization Required 11 4.611428 69.16.138.164 207.46.199.14 SIP Status: 407 Proxy Authorization Required 12 4.611574 207.46.199.14 69.16.138.164 SIP Request: ACK sip:6415551234@my.itsp.com 13 4.613772 207.46.199.15 207.46.199.14 SIP Request: ACK sip:6415551234@my.itsp.com 14 4.622070 207.46.199.15 207.46.199.14 SIP/SDP Request: INVITE sip:6415551234@my.itsp.com, with session description 15 4.623428 207.46.199.14 207.46.199.15 SIP Status: 100 trying -- your call is important to us 16 4.625164 207.46.199.14 69.16.138.164 SIP/SDP Request: INVITE sip:6415551234@my.itsp.com, with session description 17 4.648612 69.16.138.164 207.46.199.14 SIP Status: 100 Trying 18 7.101641 69.16.138.164 207.46.199.14 SIP Status: 180 Ringing 19 7.101844 207.46.199.14 207.46.199.15 SIP Status: 180 Ringing 20 7.601009 69.16.138.164 207.46.199.14 SIP Status: 180 Ringing 21 7.601206 207.46.199.14 207.46.199.15 SIP Status: 180 Ringing 22 8.859386 69.16.138.164 207.46.199.14 SIP Status: 180 Ringing 23 8.859577 207.46.199.14 207.46.199.15 SIP Status: 180 Ringing .... ....
My config:
# ----------- global configuration parameters ------------------------
fork=no log_stderror=yes debug=7
check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=1 fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
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", 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") {
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(); 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(); }; }