[OpenSIPS-Users] Opensips 1.9 as registrar and rtp_proxy / firewall

[OCEANET - Cédric BASSAGET]

What do you have in usrloc ?

Cédric
Le 10/04/2013 18:19, Andrei Grav a écrit :
> Hi,
> I have 2 NIC's but the secodn one is not used
> The opensips is listten only on eth0 with public IP and default route
> So the rtp is not in bridge mode
>
> Regards,
> Andrei
>
>
> On Wed, Apr 10, 2013 at 7:14 PM, OCEANET - Cédric BASSAGET 
> <cedric at oceanet.com <mailto:cedric at oceanet.com>> wrote:
>
>     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:+0080080000 at myopensips.com:5060
>>     <http://sip:+0080080000@myopensips.com:5060> SIP/2.0.
>>     Via: SIP/2.0/UDP
>>     89.xxx.xxx.xxx:47054;rport;branch=z9hG4bKPjhaz3VKGysNj2NSHtuKad0rIMpg3pUdt9.
>>     Max-Forwards: 70.
>>     From: <sip:50105 at myopensips.com
>>     <mailto:sip%3A50105 at myopensips.com>>;tag=szMaKKCRwqx5OGjUCwNS7F4X1ZDQpr8F.
>>     To: <sip:+0080080000 at myopensips.com
>>     <mailto:sip%3A%2B0080080000 at myopensips.com>>.
>>     Contact: <sip:50105 at 89.xxx.xxx.xxx: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
>>
>>
>>     _______________________________________________
>>     Users mailing list
>>     Users at lists.opensips.org  <mailto:Users at lists.opensips.org>
>>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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