[Serusers] Another question regarding Asterisk & SER
Curt Moore
tgrman21 at gmail.com
Sat Jul 24 08:12:32 CEST 2004
As a preface, my setup is somewhat unique; if anyone has a working SER
configuration for this scenario, or is willing to help come up with
one, I would be willing to compensate you for your efforts. If you're
interested, please contact me off list for details.
At present, I'm using Asterisk to run a small VoIP network with great
success. My focus is now to look at handling things such as NAT and
scalability/reliability. This where I believe SER will be very
beneficial as it handles NAT and scaling very well.
My desired setup is as follows (the only thing behind a NAT are phones):
Phone(s) --> [NAT] --> SER --> Asterisk(FS) --> Asterisk (PSTN)
All phones behind the NAT have their outbound proxy set to SER and
their regular proxy set to Asterisk(FS). SER then proxies outbound
traffic from the phone to the Asterisk(FS) server acting as the
"feature server" ie. VM, conferencing, etc.
The feature server then routes traffic according to the dialplan in
Asterisk. If the call is destined for the PSTN, the dialplan logic in
Asterisk(FS) sends an INVITE to Asterisk(PSTN) which tells
Asterisk(PSTN) to reINVITE the Phone at the public IP of the NAT and
also tells the phone, via SER, to reINVITE Asterisk(PSTN) at its
public IP. The media streams should then flow between the phone and
Asterisk(PSTN) while both SER and Asterisk(FS) are out of the picture
until the call ends and BYEs are processed.
If a call is destined for another SIP device, the process is the same
except that Asterisk(FS) forwards traffic back to the address of the
phone, which is SER since SER is acting as the proxy for all SIP
devices. SER then forwards the traffic back to the NATted phone as
appropriate.
Amazingly, I already have what I believe to be an SER config which
correctly routes all of the SIP/SDP traffic as desired. My current
issue is that during a call intitated by the NATted phone, the phone
and Asterisk(PSTN) have reinvited and are sending RTP to each other
but the phone behind the NAT cannot hear any of the audio being sent
by Asterisk(PSTN) while the phone on the PSTN can hear audio being
sent by the NATted phone; one way audio. What is really odd about
this is that when the PSTN user first answers the call, approximately
1-2 seconds of audio is heard on the NATted phone from the PSTN user
but then no more is heard for the remainder of the call.
For me, I see this as some sort of RTP issue as the first second of
voice RTP traffic is doing exactly what is expected between the phone
and Asterisk(PTSN). I've verified the SRC and DST of the RTP traffic
with Ethereal. I've also tried force_rtp_proxy and rtpproxy with no
success.
It is my understanding that my desired setup is very similar to that
of others such as FWD (Free World Dialup), et al.
My SER config follows. It is based on an Asterisk+SER configuration
someone posted earlier so please point out anything you see which may
not be correct.
I'll also add that this configuration works flawlessly whenever there
is only a phone and the 2 Asterisk servers; ie. no SER and no NAT.
All reINVITES take place as they should and all media is heard as it
should be.
Thanks in advance!
-Curt
#alias=" mydomian.com "
#Alias="192.168.10.100" #ser
#Alias="192.168.10.120" #Asterisk
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"
fifo_mode=438
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
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/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"
loadmodule "/usr/lib/ser/modules/acc.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 config),
# 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)
# -- acc params --
# set reporting log level
modparam("acc", "db_url", "sql://ser:heslo@localhost/ser")
modparam("acc", "db_flag", 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")) {
if(search("^Contact:(.*)192\.168(.*)")) {
log(1,"We're natted\n");
# 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
save_noreply("location");
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
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_noreply("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
# record_route();
if (isflagset(6)) {
fix_nated_sdp("1");
force_rtp_proxy();
t_on_reply("1");
}
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();
fix_nated_sdp("1");
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")) {
} else if (search("Contact:.*192\.168.*")) {
fix_nated_contact();
};
}
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