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<tt>Hello Bogdan,<br>
<br>
As your script does not generate the 488, it is for sure you get
it from the endpoints. As you already figured out, usually a 488
means a codec in-compatibility between the 2 end points. <br>
<br>
As this is completely unrelated to the proxy (not even the usage
of rtpproxy may break the codec stuff), and as it is completely
random, I would suggest taking this issue to the pjsua guys - in
what conditions the client fires the 488 - it is only based on the
codecs ? <br>
<br>
BTW, are you sure that in all your case, the 2 end points do
advertise the same list of codecs (like in all the calls ?)<br>
<br>
Best regards,<br>
</tt>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
<br>
On 06/06/2013 08:26 PM, Bogdan Chifor wrote:
<blockquote
cite="mid:CAGP9oUUfW2VAek_mmcUaUROc2y1J07ARiBCzeoqHFyCfD4a0-w@mail.gmail.com"
type="cite">
<div dir="ltr">
<div>
<div>Hello,<br>
<br>
</div>
I am using OpenSIPs as a SIP server solution.I just finished a
server configuration which permits NAT traversal (I used RTP
proxy)(on Debian).<br>
</div>
<div>As a SIP client application I am using CSipSimple(Android)
(which uses PJSUA as backend solution).<br>
</div>
<div>The setup works fine and the NAT traversal problem is
solved. (I am able to call from a 3G network provider to my
wireless LAN-it uses NAT on both sides and it works great).<br>
</div>
<div>However sometimes I am getting the 488 Not acceptable here
error. Sometimes this error appears even when both of the
phones are in the same LAN.<br>
<br>
</div>
<div>Also this error appeared when I called from the 3G network
operator to my wireless LAN.(when I called from my wireless
LAN to the 3G operator it worked fine).<br>
<br>
</div>
<div>The conclusion is that I cannot replicate this error every
time.<br>
<br>
I know that is a SDP issue.<br>
<br>
</div>
<div>Here is my opensips.cfg</div>
<div><br>
#<br>
# $Id: nathelper.cfg 9345 2012-10-18 20:24:22Z osas $<br>
#<br>
# simple quick-start config script including nathelper support<br>
<br>
# This default script includes nathelper support. To make it
work<br>
# you will also have to install Maxim's RTP proxy. The proxy
is enforced<br>
# if one of the parties is behind a NAT.<br>
#<br>
# If you have an endpoing in the public internet which is
known to<br>
# support symmetric RTP (Cisco PSTN gateway or voicemail, for
example),<br>
# then you don't have to force RTP proxy. If you don't want to
enforce<br>
# RTP proxy for some destinations than simply use t_relay()
instead of<br>
# route(1)<br>
#<br>
# Sections marked with !! Nathelper contain modifications for
nathelper<br>
#<br>
# NOTE !! This config is EXPERIMENTAL !<br>
#<br>
# ----------- global configuration parameters
------------------------<br>
<br>
debug=5<br>
log_stderror=no<br>
log_facility=LOG_LOCAL0<br>
#log_name=opensips.log<br>
<br>
fork=yes<br>
children=4<br>
<br>
/* uncomment the following lines to enable debugging */<br>
#debug=6<br>
#fork=no<br>
#log_stderror=yes<br>
<br>
<br>
check_via=no # (cmd. line: -v)<br>
dns=no # (cmd. line: -r)<br>
rev_dns=no # (cmd. line: -R)<br>
<br>
#port=5060<br>
listen=udp:<a moz-do-not-send="true"
href="http://195.95.167.214:8899">195.95.167.214:8899</a><br>
#advertised_address="195.95.167.194"<br>
#advertised_port=5060<br>
port=8899<br>
<br>
<br>
children=4<br>
<br>
disable_tcp=yes<br>
<br>
alias=<a moz-do-not-send="true" href="http://voip.certsign.ro">voip.certsign.ro</a><br>
alias=195.95.167.214<br>
#alias=192.168.185.26<br>
# ------------------ module loading
----------------------------------<br>
<br>
#set module path<br>
mpath="/usr/lib/opensips/modules/"<br>
<br>
# Uncomment this if you want to use SQL database<br>
loadmodule "db_mysql.so"<br>
<br>
loadmodule "sl.so"<br>
loadmodule "tm.so"<br>
loadmodule "signaling.so"<br>
loadmodule "rr.so"<br>
loadmodule "maxfwd.so"<br>
loadmodule "usrloc.so"<br>
loadmodule "registrar.so"<br>
loadmodule "textops.so"<br>
loadmodule "mi_fifo.so"<br>
loadmodule "sipmsgops.so"<br>
loadmodule "dialog.so"<br>
loadmodule "avpops.so"<br>
loadmodule "domain.so"<br>
#loadmodule "xlog.so"<br>
loadmodule "acc.so"<br>
<br>
# Uncomment this if you want digest authentication<br>
# db_mysql.so must be loaded !<br>
loadmodule "auth.so"<br>
loadmodule "auth_db.so"<br>
<br>
# !! Nathelper<br>
loadmodule "nathelper.so"<br>
loadmodule "rtpproxy.so"<br>
<br>
# ----------------- setting module-specific parameters
---------------<br>
<br>
# -- mi_fifo params --<br>
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")<br>
<br>
# -- usrloc params --<br>
modparam("usrloc", "db_mode", 2)<br>
modparam("usrloc", "db_url",
"mysql://opensips:qwe123@localhost/opensips")<br>
<br>
<br>
# -- auth params --<br>
# Uncomment if you are using auth module<br>
modparam("auth_db", "calculate_ha1", yes)<br>
#<br>
# If you set "calculate_ha1" parameter to yes (which true in
this config),<br>
# uncomment also the following parameter)<br>
modparam("auth_db", "password_column", "password")<br>
modparam("auth_db", "db_url",
"mysql://opensips:qwe123@localhost/opensips")<br>
modparam("auth_db", "load_credentials", "")<br>
<br>
db_default_url="mysql://opensips:qwe123@localhost/opensips"<br>
<br>
<br>
#<br>
# !! Nathelper<br>
#<br>
modparam("usrloc", "nat_bflag", 6)<br>
modparam("nathelper", "ping_nated_only", 1) # Ping only
clients behind NAT<br>
modparam("nathelper", "sipping_bflag", 8)<br>
modparam("nathelper", "received_avp", "$avp(i:801)")<br>
<br>
# RTPProxy setup<br>
#modparam("nathelper", "rtpproxy_sock", "udp:<a
moz-do-not-send="true" href="http://127.0.0.1:7789">127.0.0.1:7789</a>")<br>
modparam("nathelper", "force_socket", "udp:<a
moz-do-not-send="true" href="http://127.0.0.1:7789">127.0.0.1:7789</a>")<br>
<br>
modparam("rtpproxy", "rtpproxy_sock", "udp:<a
moz-do-not-send="true" href="http://127.0.0.1:7789">127.0.0.1:7789</a>")<br>
modparam("rtpproxy", "rtpproxy_autobridge", 1)<br>
modparam("rtpproxy", "rtpproxy_timeout", "0.5")<br>
modparam("rtpproxy", "rtpproxy_retr", 3)<br>
<br>
<br>
#<br>
# ------- dialog --------<br>
#**<br>
modparam("dialog", "db_mode", 1)<br>
modparam("dialog", "db_update_period", 30)<br>
#modparam("dialog", "dlg_flag", 4)<br>
modparam("dialog", "dlg_match_mode", 1)<br>
<br>
#<br>
# --> avpops params -----<br>
#**<br>
modparam("avpops", "avp_table", "usr_preferences")<br>
modparam("avpops", "use_domain", 1)<br>
<br>
<br>
<br>
# ************<br>
# ----- presence params -----<br>
/* uncomment the following lines if you want to enable
presence */<br>
#modparam("presencepresence_xml", "db_url",<br>
# "mysql://opensips:opensipsrw@localhost/opensips")<br>
#modparam("presence_xml", "force_active", 1)<br>
#modparam("presence", "server_address", <a class="moz-txt-link-rfc2396E" href="sip:localhost:5060">"sip:localhost:5060"</a>)<br>
<br>
<br>
# ------------------------- request routing logic
-------------------<br>
<br>
# main routing logic<br>
<br>
route{<br>
# initial sanity checks -- messages with<br>
# max_forwards==0, or excessively long requests<br>
if (!mf_process_maxfwd_header("10")) {<br>
sl_send_reply("483","Too Many Hops");<br>
exit;<br>
};<br>
if (msg:len >= 2048 ) {<br>
sl_send_reply("513", "Message too big");<br>
exit;<br>
};<br>
<br>
# !! Nathelper<br>
# Special handling for NATed clients; first, NAT test
is<br>
# executed: it looks for via!=received and RFC1918
addresses<br>
# in Contact (may fail if line-folding is used); also,<br>
# the received test should, if completed, should check
all<br>
# vias for rpesence of received<br>
#if (nat_uac_test("3")) {<br>
# Allow RR-ed requests, as these may indicate
that<br>
# a NAT-enabled proxy takes care of it; unless
it is<br>
# a REGISTER<br>
<br>
if (is_method("REGISTER") ||
!is_present_hf("Record-Route")) {<br>
log("LOG:Someone trying to register
from private IP, rewriting\n");<br>
# This will work only for user agents
that support symmetric<br>
# communication. We tested quite many
of them and majority is<br>
# smart enough to be symmetric. In
some phones it takes a<br>
# configuration option. With Cisco
7960, it is called<br>
# NAT_Enable=Yes, with kphone it is
called "symmetric media" and<br>
# "symmetric signalling".<br>
<br>
# Rewrite contact with source IP of
signalling<br>
fix_nated_contact();<br>
if ( is_method("INVITE") ) {<br>
log("@DEBUG:FIX SDP");<br>
fix_nated_sdp("2"); # Add
direction=active to SDP<br>
};<br>
force_rport(); # Add rport parameter
to topmost Via<br>
setbflag(6); # Mark as NATed<br>
<br>
# if you want sip nat pinging<br>
# setbflag(8);<br>
};<br>
#};<br>
<br>
# subsequent messages withing a dialog should take the<br>
# path determined by record-routing<br>
if (loose_route()) {<br>
# mark routing logic in request<br>
append_hf("P-hint: rr-enforced\r\n");<br>
route(1);<br>
exit;<br>
};<br>
# we record-route all messages -- to make sure that<br>
# subsequent messages will go through our proxy;
that's<br>
# particularly good if upstream and downstream
entities<br>
# use different transport protocol<br>
if (!is_method("REGISTER"))<br>
record_route();<br>
<br>
if (!uri==myself) {<br>
# mark routing logic in request<br>
append_hf("P-hint: outbound\r\n");<br>
route(1);<br>
exit;<br>
};<br>
<br>
# if the request is for other domain use UsrLoc<br>
# (in case, it does not work, use the following
command<br>
# with proper names and addresses in it)<br>
if (uri==myself) {<br>
<br>
if (is_method("REGISTER")) {<br>
<br>
# Uncomment this if you want to use
digest authentication<br>
#if (!www_authorize("<a
moz-do-not-send="true" href="http://siphub.org">siphub.org</a>",
"subscriber")) {<br>
# www_challenge("<a
moz-do-not-send="true" href="http://siphub.org">siphub.org</a>",
"0");<br>
# return;<br>
#};<br>
<br>
save("location");<br>
exit;<br>
};<br>
<br>
lookup("aliases");<br>
if (!uri==myself) {<br>
append_hf("P-hint: outbound
alias\r\n");<br>
route(1);<br>
exit;<br>
};<br>
<br>
# native SIP destinations are handled using
our USRLOC DB<br>
if (!lookup("location")) {<br>
sl_send_reply("404", "Not Found");<br>
exit;<br>
};<br>
};<br>
append_hf("P-hint: usrloc applied\r\n");<br>
log("@DEBUG:LINE BEFORE ROUTE[1]");<br>
route(1);<br>
}<br>
<br>
route[1]<br>
{<br>
# !! Nathelper<br>
if
(uri=~<a class="moz-txt-link-rfc2396E" href="mailto:[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)">"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)"</a>
&& !search("^Route:")){<br>
sl_send_reply("479", "We don't forward to
private IP addresses");<br>
exit;<br>
};<br>
<br>
# if client or server know to be behind a NAT, enable
relay<br>
if (isbflagset(6)){<br>
<br>
if (is_method("INVITE")) {<br>
log("@DEBUG:INVITE METHOD");<br>
if (has_body("application/sdp")) {<br>
log("@DEBUG:application/sdp");<br>
if
(rtpproxy_offer("195.95.167.214")) {<br>
t_on_reply("2");<br>
log("@DEBUG:RTP PROXY
OFFER");<br>
}<br>
} else {<br>
t_on_reply("3");<br>
}<br>
}<br>
if (is_method("ACK") &&
has_body("application/sdp"))<br>
rtpproxy_answer();<br>
};<br>
<br>
# NAT processing of replies; apply to all transactions
(for example,<br>
# re-INVITEs from public to private UA are hard to
identify as<br>
# NATed at the moment of request processing); look at
replies<br>
t_on_reply("1");<br>
<br>
# send it out now; use stateful forwarding as it works
reliably<br>
# even for UDP2TCP<br>
if (!t_relay()) {<br>
sl_reply_error();<br>
};<br>
}<br>
<br>
# !! Nathelper<br>
onreply_route[1] {<br>
# NATed transaction ?<br>
if (isbflagset(6) && status =~
"(183)|2[0-9][0-9]") {<br>
fix_nated_contact();<br>
rtpproxy_answer();<br>
# otherwise, is it a transaction behind a NAT and we
did not<br>
# know at time of request processing ? (RFC1918
contacts)<br>
} else if (nat_uac_test("1")) {<br>
fix_nated_contact();<br>
};<br>
}<br>
onreply_route[2]<br>
{<br>
<br>
if (has_body("application/sdp"))<br>
rtpproxy_answer("195.95.167.214");<br>
}<br>
<br>
onreply_route[3]<br>
{<br>
<br>
if (has_body("application/sdp"))<br>
rtpproxy_offer("195.95.167.214");<br>
}<br>
<br>
<br>
<br>
<div>Please help me.<br>
<br>
</div>
Thank you very much,<br>
<br>
</div>
<div>Chifor Bogdan<br>
</div>
</div>
<pre wrap="">
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</blockquote>
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