[Users] nat_helper: multiple media IP address in SDP

[Nicolas Olivier]

Hi,

I've got a gateway which is only used for rounting and rtp proxying between providers and centrexes.

On reply to an INVITE, one of our provider send back a "183 Session Progress". The problem is that in the SDP block, we've got 2 media IP address and 
rtpproxy only rewrite one.

Finally, the provider establish rtp session with our gateway, and our centrex directly with the provider.

   provider                  gateway                  centrex
172.16.0.10               192.168.1.10              192.168.1.20
      RTP     ------------->   RTP      ------------>   RTP
       ^-------------------------------------------------|

So my questions are, is it possible to have multiple IP address in SDP and if so, how can I tell rtpproxy to rewrite all of them.

Coming from provider:

SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 192.168.1.10;branch=z9hG4bKdd67.a4cc2c44.0,SIP/2.0/UDP 192.168.1.20:5062;branch=z9hG4bKdd67.08f45a33.0,SIP/2.0/UDP 
192.168.1.20:5060;branch=z9hG4bK4af242b7.
From: "02" <sip:0143132445 at 192.168.1.20>;tag=as226ce7b9.
To: <sip:0123456789 at 192.168.1.20:5062>;tag=3123AAA8-20C5.
Date: Tue, 11 Apr 2006 09:10:29 GMT.
Call-ID: 079ab6663e403ff44a1107e5111b075f at 192.168.1.20.
Server: Cisco-SIPGateway/IOS-12.x.
CSeq: 102 INVITE.
Allow-Events: telephone-event.
Contact: <sip:677238#0123456789 at 172.16.0.10:5060>.
Record-Route: <sip:192.168.1.10;ftag=as226ce7b9;lr=on>,<sip:192.168.1.20:5062;ftag=as226ce7b9;lr=on>.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Content-Length: 261.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 3448 4768 IN IP4 172.16.0.10.
s=SIP Call.
c=IN IP4 172.16.0.10.
t=0 0.
m=audio 18322 RTP/AVP 18 101.
c=IN IP4 172.16.0.10.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.

Forwarded to centrex:

SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 192.168.1.20:5062;branch=z9hG4bK43a4.3e96aba3.0,SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK3213db83.
From: "02" <sip:0143132445 at 192.168.1.20>;tag=as1a2f900d.
To: <sip:0123456789 at 192.168.1.20:5062>;tag=3121D1B4-1BFD.
Date: Tue, 11 Apr 2006 09:08:28 GMT.
Call-ID: 08467c5e299ab833106517c63d3edc2e at 192.168.1.20.
Server: Cisco-SIPGateway/IOS-12.x.
CSeq: 102 INVITE.
Allow-Events: telephone-event.
Contact: <sip:677238#0123456789 at 172.16.0.10:5060>.
Record-Route: <sip:192.168.1.10;ftag=as1a2f900d;lr=on>,<sip:192.168.1.20:5062;ftag=as1a2f900d;lr=on>.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Content-Length: 277.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 565 174 IN IP4 172.16.0.10.
s=SIP Call.
c=IN IP4 172.16.0.10.
t=0 0.
m=audio 36296 RTP/AVP 18 101.
c=IN IP4 192.168.1.10.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=nortpproxy:yes.


openser.cfg

(...)

  onreply_route[1] {
          if (status =~ "(180)|(183)|2[0-9][0-9]") {
                  fix_nated_contact();
                  if (!search("^Content-Length:[ ]*0")) {
                          force_rtp_proxy();
                  };
          } else if (nat_uac_test("1")) {
                  fix_nated_contact();
          };
  }

(...)

Best regards,
Nicolas Olivier