[Serusers] ZyXel P2000w wlan SIP UA's
mike
mseppane at dc.turkuamk.fi
Fri May 28 10:06:53 CEST 2004
Maybe I should be more specific with the question.
At the moment I am only testing IP-to IP without PSTN connectivity. I'm
still waiting for an FXO module for my channelbank Adtran Atlas 550-series
(anyone tested this?)which I plan on integrating w/ Asterisk. . maybe.
Basically I now have SER /RTP running with a basic configuration. See
below--> I have been modifying ser.cfg lots just to experiment and learn.
I have two separate locations at my use , the other office(1) being behind
a NAT and on the other side where my research lab(2) is no NAT/firewall
with public addresses. On top of this I have a public hotspot available to
my lab which uses RADIUS/LDAP authentication with a pre-assigned login
name and password.
To get to the point then.
At office 1, When I first received the phones, tried them out behind a
D-link wireless router, so basically it was behind two nat's :( (what's
the scenario on this?) Inside the private dhcp-generated IP-space, calls
between the UA's worked fine. phones sip-settings as my SER
(sip3.dc.turkuamk.fi) for both proxy and outgoing. SER did log an incoming
request, but at the moment didn't even have NAT-helper conf.d. so the
reply was lost. Even afterwards the phones would not register to my
server. Now my research-lab doesn't have a wireless network except for the
hotspot. The Prestige finds the network and receives an IP-address, but
fails to make calls and register on both sip3.dc.turkuamk.fi server nor
iptel.org servers. To my understanding the "SPARKNET" hotspot uses also
ssl encryption, but I'm not sure. I am also a bit strange on the
authentication scenario due to the public access point. The phone has no
setting for such authentication. Ofcourse there is the SIP-setting
username and password, but that's it.
Otherwise my current configuration has worked fine for softphones,
(kphone, M$-mes.. and SCS-client)making, receiving and registering calls
and UA's using the digest authentication scheme
even behind the NAT. Sip-clients work fine on my laptop on the hotspot.
Maybe I should get another d-link wireless network... :)
thanks,
Michael Seppänen
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
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"
alias="sip3.dc.turkuamk.fi"
alias="wlan.turkuamk.fi"
alias="sip.dc.turkuamk.fi"
alias="turkuamk.fi"
alias="brage-dataware.fi"
alias="10.42.1.179"
alias="93.166.139.107"
listen=193.166.135.116
# ------------------ module loading ----------------------------------
loadmodule "/usr/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/auth.so"
loadmodule "/usr/lib/ser/modules/auth_db.so"
loadmodule "/usr/lib/ser/modules/acc.so"
loadmodule "/usr/lib/ser/modules/exec.so"
loadmodule "/usr/lib/ser/modules/group.so"
#loadmodule "/usr/lib/ser/modules/msilo.so"
loadmodule "/usr/lib/ser/modules/textops.so"
#loadmodule "/usr/lib/ser/modules/uri.so"
loadmodule "/usr/lib/ser/modules/nathelper.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 2)
# -- auth params --
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
# -- rr params --
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("", "subscriber")) {
www_challenge("", "0");
break;
};
save("location");
break;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
lookup("aliases");
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
};
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();
};
}
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