[Users] nat_helper: multiple media IP address in SDP
Nicolas Olivier
nolivier at alphalink.fr
Tue Apr 11 16:51:22 CEST 2006
Ok, I may have a look to the csv.
Thanks for the help.
regards,
Nicolas
Bogdan-Andrei Iancu wrote:
> 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
> >> >
> >> >
> >> > _______________________________________________
> >> > Users mailing list
> >> > Users at openser.org
> >> > http://openser.org/cgi-bin/mailman/listinfo/users
> >> >
> >>
> >
>
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