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

Andrei Grav andreigrav at gmail.com
Wed Apr 10 18:19:22 CEST 2013


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> 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 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>;tag=szMaKKCRwqx5OGjUCwNS7F4X1ZDQpr8F.
> To: <sip:+0080080000 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
>
>
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