11 Apr
2006
11 Apr
'06
4:51 p.m.
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@192.168.1.20>;tag=as226ce7b9.
> > To: <sip:0123456789@192.168.1.20:5062>;tag=3123AAA8-20C5.
> > Date: Tue, 11 Apr 2006 09:10:29 GMT.
> > Call-ID: 079ab6663e403ff44a1107e5111b075f(a)192.168.1.20.
> > Server: Cisco-SIPGateway/IOS-12.x.
> > CSeq: 102 INVITE.
> > Allow-Events: telephone-event.
> > Contact: <sip:677238#0123456789@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@192.168.1.20>;tag=as1a2f900d.
> > To: <sip:0123456789@192.168.1.20:5062>;tag=3121D1B4-1BFD.
> > Date: Tue, 11 Apr 2006 09:08:28 GMT.
> > Call-ID: 08467c5e299ab833106517c63d3edc2e(a)192.168.1.20.
> > Server: Cisco-SIPGateway/IOS-12.x.
> > CSeq: 102 INVITE.
> > Allow-Events: telephone-event.
> > Contact: <sip:677238#0123456789@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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