Thank you Bogdan, I solved the problem based on your idea. I'm not to sure about all because I'm at beginning of use of Opensips.... I am going to read more about it.
best regards, Andrei On Wed, Apr 10, 2013 at 7:43 PM, Bogdan-Andrei Iancu <[email protected]>wrote: > ** > Hello Andrei, > > For such a call (to a public end point), you do not actually need a media > relay as Asterisk could do Comedia (or symmetric RTP). > > The idea is to detect on OpenSIPS the presence of NAT , take care of > fixing the contact of UAC and on SDP part use fix_nated_sdp("1") (see > http://www.opensips.org/html/docs/modules/1.9.x/nathelper.html#id250434) > for force Comedia on Asterisk. No need for a media relay. > > Best regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > > On 04/10/2013 06:42 PM, Andrei Grav wrote: > > 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 SIP/2.0. > Via: SIP/2.0/UDP > 89.xxx.xxx.xxx:47054;rport;branch=z9hG4bKPjhaz3VKGysNj2NSHtuKad0rIMpg3pUdt9. > Max-Forwards: 70. > From: <sip:[email protected]>;tag=szMaKKCRwqx5OGjUCwNS7F4X1ZDQpr8F. > To: <sip:[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 > > > _______________________________________________ > Users mailing > [email protected]http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
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