Hi
Does your opensips have two network interfaces, and does rtpproxy works
in bridge mode ?
Cédric
Le 10/04/2013 17:42, Andrei Grav a écrit :
Hi,
I have installed Opensips 1.9 on debian 6 using the default config for
residential with NAT
My topology is:
UAC ------ ROUTER+FIREWALL --------[ INTERNET ]---- OPENSIPS -----
ASTERISK
UAC is behind NAT 89.xxx.xxx.xxx
Opensips and Asterisk has Public IP's
I'm facing the following issue:
The invite from UAC is announcing his audio port on 4000
Because is not NAT detected the asterisk is sending the rtp media
directly to 89.x.x.x:4000 and no using rtp_proxy
Because the firewall is blocking the RTP port 4000 coming from
Asterisk's IP there is no audio on the UAC side
IF I comment the line #if (nat_uac_test("23")) all the traffic is
going to rtp_proxy and audio is working fine.
Is there any way to solve this ?
route{
force_rport();
if (nat_uac_test("23")) {
if (is_method("REGISTER")) {
fix_nated_register();
setbflag(NAT);
} else {
fix_nated_contact();
setflag(NAT);
}
}
.....
U 89.xxx.xxx.xxx:47054 -> 193.xxx.xxx.xxx:5060
INVITE sip:[email protected]:5060
<http://sip:[email protected]:5060> SIP/2.0.
Via: SIP/2.0/UDP
89.xxx.xxx.xxx:47054;rport;branch=z9hG4bKPjhaz3VKGysNj2NSHtuKad0rIMpg3pUdt9.
Max-Forwards: 70.
From: <sip:[email protected]
<mailto:sip%[email protected]>>;tag=szMaKKCRwqx5OGjUCwNS7F4X1ZDQpr8F.
To: <sip:[email protected]
<mailto:sip%3a%[email protected]>>.
Contact: <sip:[email protected]:47054;ob>.
Call-ID: sLw5Y3pTNKyBLrk2ZS9brsL87jxN6CPZ.
CSeq: 3657 INVITE.
Route: <sip:myopensips.com:5060;transport=udp;lr>.
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS.
Supported: replaces, 100rel, timer, norefersub.
Session-Expires: 1800.
Min-SE: 90.
User-Agent: CSipSimple_GT-I9100-16/r1916.
Content-Type: application/sdp.
Content-Length: 348.
.
v=0.
o=- 3574590691 3574590691 IN IP4 89.xxx.xxx.xxx.
s=pjmedia.
c=IN IP4 89.xxx.xxx.xxx.
t=0 0.
m=audio 4000 RTP/AVP 99 0 8 101.
c=IN IP4 89.xxx.xxx.xxx.
a=rtcp:4001 IN IP4 89.xxx.xxx.xxx.
a=sendrecv.
a=rtpmap:99 SILK/24000.
Regards,
Andrei
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