[Users] nat_helper: multiple media IP address in SDP

[Bogdan-Andrei Iancu]

Hi,

Nicolas Olivier wrote:

>
> Hi Bogdan,
>
> Ok, I understand now. But I still encounter the problem because:
> - rtpproxy only rewrites the c= from media part (but it should be fine 
> as you said) despite what a quick look in the rtpproxy code comments 
> say ("We have to change ports in m-lines, and also change IP addresses 
> in c-lines which can be placed either in session header (fallback for 
> all medias) or media description.")

yes, the nathelper will change the c= from session header only if it 
finds a media section without a local c= (which means the default c= 
from session hdr will be used).

> - the centrex (which is an asterisk by the way) take only into account 
> the c= from the session part, not the one from media part

in the CVS devel there is a flag that force also changing of session c= :
    http://openser.org/docs/modules/1.1.x/nathelper.html#AEN275 , the 
"c" flag

regards,
bogdan

>
>
>
> regards,
> Nicolas
>
> Bogdan-Andrei Iancu wrote:
>
>> 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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>>  > Users at openser.org
>>  > http://openser.org/cgi-bin/mailman/listinfo/users
>>  >
>>
>