Hi,
I am using the following config with Opensips and having a problem with one way
audio. When connecting the softphone directly to asterisk that runs on the same
machine audio passes without any problems. Firewalls are all open and
Zoiper/Snom phones connect without issue.
If anyone could offer some advice it would be much appreciated as I'm tearing
my hear out :-)
# --- global configuration parameters
debug=3 # debug level (cmd line: -dd)
fork=yes
log_stderror=yes # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
listen=udp:IP ADDRESS:5060
children=4
# -- module loading --
#set module path
mpath=/usr/local/lib64/opensips/modules/
# Uncomment this if you want to use SQL database
loadmodule db_mysql.so
loadmodule sl.so
loadmodule tm.so
loadmodule signaling.so
loadmodule rr.so
loadmodule maxfwd.so
loadmodule usrloc.so
loadmodule registrar.so
loadmodule textops.so
loadmodule mi_fifo.so
loadmodule mediaproxy.so
loadmodule xlog.so
loadmodule dialog.so
loadmodule load_balancer.so
loadmodule mi_datagram.so
# Uncomment this if you want digest authentication
# db_mysql.so must be loaded !
#loadmodule auth.so
#loadmodule auth_db.so
# !! Nathelper
loadmodule nathelper.so
# - setting module-specific parameters ---
# -- mi_fifo params --
modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)
modparam(mi_datagram, socket_name, /tmp/opensips.sock)
modparam(mi_datagram, unix_socket_mode, 0600)
modparam(mi_datagram, socket_timeout, 2000)
# -- 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)
# !! Nathelper
modparam(usrloc,nat_bflag,6)
modparam(nathelper,sipping_bflag,8)
modparam(nathelper, ping_nated_only, 1) # Ping only clients behind NAT
# -- MediaProxy --
modparam(mediaproxy, mediaproxy_socket,
/var/run/mediaproxy/dispatcher.sock)
modparam(mediaproxy, mediaproxy_timeout, 500)
modparam(dialog, dlg_flag, 13)
modparam(dialog, db_mode, 1)
modparam(dialog, db_url, DB URL)
modparam(load_balancer, db_url,DB URL)
# - 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);
exit;
};
if (msg:len= 2048 ) {
sl_send_reply(513, Message too big);
exit;
};
# !! 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 (is_method(REGISTER) || !is_present_hf(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.
# Rewrite contact with source IP of signalling
fix_nated_contact();
if ( is_method(INVITE) ) {
fix_nated_sdp(1); # Add direction=active to SDP
};
force_rport(); # Add rport parameter to topmost Via
setbflag(6);# Mark as NATed
# if you want sip nat pinging
# setbflag(8);
};
};
# 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);
exit;
};
# 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 (!is_method(REGISTER))
record_route();
if (!uri==myself) {
# mark routing logic in request
append_hf(P-hint: outbound\r\n);
route(1);
exit;
};
# if the request is for other domain use UsrLoc
