31 Jan
2005
31 Jan
'05
5:26 p.m.
Hi,
I have the exact same problem as described in the below email which I
found on the serusers mailing archives. That is clients behind the
same nat can ring each other and transmit audio but if a client tries
to ring a client behind another nat no audio is transmitted.
According to the below email this is because the sdp is not rewritten
correctly. The below solution mentions calling force_rtpproxy but
that function is called in the original config (see below)...Is there
a specific plce that it should be invoked?
Thank you,
Aisling.
You have to call force_rtpproxy -> this will rewrite the IP address
and
port with the one of the rtpproxy.
klaus
Sun Yen-Rong (Dan) wrote:
enforce
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Hi Klaus,
Firstly, thanks for your help. I used ethereal to capture the SIP
messages, and found out that the SDP in the forwarded INIVITE and
200 OK
message are not rewritten correctly. As a result, the
client with a
public IP address is sending RTP packets to the private IP address
of
the other client who is behind a NAT. One seruser
suggested me to
use
fix_nated_sdp("3") instead of
fix_nated_sdp("1"), however, it didn't
solve my problem. Do I also need to add a parameter to other
nathelper
functions as well?
Regards,
Dan.
-----Original Message-----
From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at]
Sent: Thursday, May 06, 2004 9:45 PM
To: Sun Yen-Rong (Dan)
Cc: serusers at iptel.org
Subject: Re: [Serusers] no audio problem if clients behind NATs
use a packet sniffer (ethereal or ngrep) and watch the SIP messages
at
the SIP proxy. Take a look at the sdp in the forwarded
INVITE and
200 OK
message and verify that the IP address and port in the SDP are
rewritten
correctly and points to the rtpproxy.
regards,
klaus
Sun Yen-Rong (Dan) wrote:
my
Hi all,
I am using ser 0.8.12 with nathelper and rtpproxy. When I tried to
make
a call between two clients which are behind the
same NAT, everything
work fine. However, when I try to make a call between clients which
are
>behind different NATs, niether client can hear each other's audio. configuration
file is shown in the end of this message. Can someone
help
me, please?
Thanks,
Dan.
#
# $Id: nathelper.cfg,v 1.1.2.1 2003/11/24 14:47:18 janakj 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 ># 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)
>
>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
----------------------------------
>",#
># $Id: nathelper.cfg,v 1.2 2003/04/15 20:35:29 jiri Exp $
>#
># example script showing use of nathelper module
># (incomplete for sake of brevity)
>#
>
># ----------- global configuration parameters
------------------------
>
># ------------------ 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") {
# Uncomment this if you want to use digest authentication
# if (!www_authorize("iptel.org", "subscriber")) {
# www_challenge("iptel.org", "0");
# break;
# };
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();
> 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();
> };
>}