[Serusers] Duplicate SIP messages?
Tim Pushor
timp at crossthread.com
Sun Jun 26 20:46:13 CEST 2005
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 at 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 at my.itsp.com, with session description
6 4.578979 207.46.199.14 69.16.138.164 SIP/SDP
Request: INVITE sip:6415551234 at 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 at 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 at my.itsp.com
13 4.613772 207.46.199.15 207.46.199.14
SIP Request: ACK sip:6415551234 at my.itsp.com
14 4.622070 207.46.199.15 207.46.199.14
SIP/SDP Request: INVITE sip:6415551234 at 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 at 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();
};
}
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