31 Jul
2007
31 Jul
'07
11:04 a.m.
In normal cases, as mentioned earlier, removing force_rtp_proxy();
should do the trick. Otherwise, can u post somewhere your INVITES
coming in and out from the server?
DanB
On 7/31/07, Nhadie Ramos <nhadie(a)tbgi.net.ph> wrote:
Hi Dan,
My pstn gateway is on public IP, my SIP server is also on public IP.
most clients are behind NAT, using DSL routers, STUNS are configured on
the PAP2's
This is a part of the config, what can i change to not use rtpproxy, i
read somewhere to use fix_nated_sdp("3"),
i tried that but callers cannot hear the other side
if (nat_uac_test("19")) {
if (method=="REGISTER" || !search("^Record-Route:"))
{
log("LOG: Someone trying to register from
private IP, rewriting\n");
fix_nated_contact();
if (method == "INVITE") {
fix_nated_sdp("1");
};
force_rport();
setflag(6);
};
};
if ((uri=~"^(sip:)?00*@([a-z]+\.)?mydomain\.com")
|| (uri=~"^(sip:)?00*@202\.202\.202\.202")) {
if (isflagset(8)) {xlog("L_INFO", "Calls
to PSTN\n");};
if (isflagset(6)) {route(1);};
strip(2);
if (isflagset(8)) {xlog("L_INFO",
"Retail Server\n");};
prefix("12345#");
rewritehostport("203.203.203.203:5060");
consume_credentials();
t_relay_to_udp("203.203.203.203",
"5060");
break;
}
route[1] {
xlog("L_INFO", "User-Agent behind NAT\n");
force_rtp_proxy();
if (method=="INVITE") {
t_on_reply("1");
};
append_hf("P-Behind-NAT: Yes\r\n");
break;
}
onreply_route[1] {
if (status =~ "(180)|(183)|2[0-9][0-9]") {
fix_nated_contact();
if (!(search("^Content-Length:\ 0"))) {
force_rtp_proxy();
};
};
}
Regards,
Nhadie
Dan-Cristian Bogos wrote:
Hi there Nhadie,
if you use STUN or any other way of detecting the public IP (eg. ICE)
and u do it properly, u can configure OpenSER without rtpproxy.
Just make sure u have proper connection IP in your SDP (if your
devices are on PUBLIC IP make sure that is public as well).
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From the configuration point of view, u need to
remove the lines with
force_rtpproxy() and unforce_rtpproxy() and u will have no
media proxy
support then.
Let me know if u need any additional info.
Cheers,
DanB
On 7/31/07, Nhadie Ramos <nhadie(a)tbgi.net.ph> wrote:
Hi All,
Has anyone tried nathelper but not using rtpproxy, my server is on a
location with limited bandwidth so
i'd like to be able to just have the sip messages to the server but not
the media. Do i need all phones configured
to use STUN if the SIP server is configured that way?
Thank You in Advanced
Regards,
Nhadie
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