6 Aug
2013
6 Aug
'13
7:15 a.m.
Ok thanks,
All fine except the rtpproxy_manage function is just out in the open and
since you're in bridging mode you need to realize that your kamailio may
receive calls from the WAN Interface or from the Asterisks on LAN
interface. So if you're to bridge the RTPs at the proxy then how can
Kamailio detect that which combination of flags to use in rtpproxy_manage()
function?
As an example if you replace your rtpproxy_manage(); with
rtpproxy_manage("ei"); and make a call from SIP user to 777 in asterisk
dialplan where 777 is echo()
exten => 777,1,Answer()
same => n,Echo()
Take a sip trace now and see that you get the desired LAN IP of kamailio
coming in on Asterisk ? revert the "ei" to "ie" if not.
This simple test should get you a two way audio established for any call
from Internet to the Asterisk via Kamailio.
The Next step is to figure out that your call is coming from Asterisk and
wants to go out from kamailio's WAN interface. use inverse flag combination
here and let the call go out.
I hope it works for you now.
--
Sammy
On Tue, Aug 6, 2013 at 4:06 AM, Alexandr Usov <blessendor(a)gmail.com> wrote:
ps -ef | grep rtpproxy
kamailio 15853 1 0 12:41 ? 00:00:01 /usr/sbin/rtpproxy -u
kamailio -l 1.1.1.1 2.2.2.2 -s unix:/var/run/rtpproxy.sock
netstat -pln|grep rtpproxy
unix 2 [ ACC ] STREAM LISTENING 238561268
15853/rtpproxy /var/run/rtpproxy.sock
I'm trying rtpproxy_manage("ei"); rtpproxy_manage("ie");
rtpproxy_manage(); - the same if rtpproxy in bridged mode.
Config cuts:
#!define WITH_NAT
...
#!ifdef WITH_NAT
# ----- rtpproxy params -----
# modparam("rtpproxy", "rtpproxy_sock",
"udp:127.0.0.1:12221")
modparam("rtpproxy", "rtpproxy_sock",
"unix:/var/run/rtpproxy.sock")
# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from",
"sip:pinger@sip.myrealhostdomain.com
")
# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp",
"$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif
....
# Handle SIP registrations
route[REGISTRAR] {
if (is_method("REGISTER"))
{
if(isflagset(FLT_NATS))
{
setbflag(FLB_NATB);
# uncomment next line to do SIP NAT pinging
setbflag(FLB_NATSIPPING);
}
if (!save("location"))
sl_reply_error();
exit;
}
}
...
# Caller NAT detection route
route[NATDETECT] {
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(FLT_NATS);
}
#!endif
return;
}
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage();
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
2013/8/6 SamyGo <govoiper(a)gmail.com>
Please check the rtpproxy function and paste the
way it is written in
your configuration file. Share the output of "ps -ef | grep rtpproxy" and
"netstat -pln|grep rtpproxy"
--
Sammy
On Tue, Aug 6, 2013 at 3:55 AM, Alexandr Usov <blessendor(a)gmail.com>wrote;wrote:
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Note:
Asrterisk LAN IP real 192.168.144.101 but must be 2.2.2.101 in this
described network (I am missed to change before copy-pasting here).
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