[Users] nat_helper: multiple media IP address in SDP

[Bogdan-Andrei Iancu]

Hi Nicolas,

it;s perfectly ok - see the SDP RFC : an SDP may contain a default c= in 
the session part; each media section (m=) may contain an ip (c=); if it 
doesn't the session c= will be used.

regards,
bogdan

Nicolas Olivier wrote:

>
> 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
>
>
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