[OpenSIPS-Users] invites from registrar through opensips and rtpproxy

Razvan Crainea razvancrainea at opensips.org
Thu Sep 22 10:20:14 CEST 2011


Hi Alex,

According to your scenario, there is a late negotiation (SDP is 
advertised in 200OK and ACK). I see there is 'rtpproxy_offer' called for 
a 200OK, but I can't see any 'rtpproxy_answer' for an ACK message. 
Perhaps that's why the SDP in the ACK remains the same and I don't think 
it is a correct behaviour.

Regards,

--
Răzvan Crainea
OpenSIPS Developer


On 21.09.2011 19:19, hart323 wrote:
> Hi All!
>
> I have the following type of problem.
>
> 1) Registrar send INVITE to the phone1 and phone2 (through Opensips)
> 2) Phone1 reply with OK/SDP1 , phone2 reply with OK/SDP2 (through Opensips)
> 3) Opensips modifies SDP info (IP to RTPPROXY IP and port to some random
> port)
> 4) Registrar send ACK/SDP1 to phone2 and ACK/SDP2 to phone1 (through
> Opensips)
> 5) Opensips only relay
>
> The result is one direction voice.
>
> Here is the following config:
>
>          if (is_method("INVITE") || (is_method("ACK")&&  src_ip !=
> "REGISTRAR_IP")) {
>                  if (rtpproxy_offer("f")) {
>                          t_on_reply("1");
>                  } else {
>                          t_on_reply("2");
>                  }
>          }
>          if (is_method("ACK")&&  src_ip == "REGISTRAR_IP") {
>          }
>
> onreply_route[1] {
>          if(status =~ "180|200"&&  search("Content-Type: application/sdp")) {
>                  if (nat_uac_test("1")) {
>                          fix_nated_contact();
>                  }
>                  rtpproxy_answer("f");
>          }
> }
>
> onreply_route[2] {
>          if(status =~ "180|200"&&  search("Content-Type: application/sdp")) {
>                  if (nat_uac_test("1")) {
>                          fix_nated_contact();
>                  }
>                  rtpproxy_offer("f");
>          }
> }
>
> Here is the debug from rtpproxy when the sessions timeout:
>
> INFO:process_rtp: session timeout
> INFO:remove_session: RTP stats: 29 in from callee, 25 in from caller, 54
> relayed, 0 dropped
> INFO:remove_session: RTCP stats: 1 in from callee, 9 in from caller, 10
> relayed, 0 dropped
> INFO:remove_session: session on ports 39742/*45048 *is cleaned up
> INFO:process_rtp: session timeout
> INFO:remove_session: RTP stats: 867 in from callee, 239 in from caller, 1106
> relayed, 0 dropped
> INFO:remove_session: RTCP stats: 14 in from callee, 2 in from caller, 16
> relayed, 0 dropped
> INFO:remove_session: session on ports *42616*/38680 is cleaned up
>
> In the packet dumps bolded values are the ports in modified SDP packet(
> after rtpproxy_offer).
> Why it is different order of these values? phone2 send to 42616 and goes out
> from 38680 (working), but phone1 tries to send to 45048 and opensips drop
> this.
>
> Any help?! Plz!
>
> Best regards,
> Alex
>
> --
> View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/invites-from-registrar-through-opensips-and-rtpproxy-tp6816755p6816755.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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