11 Apr
2006
11 Apr
'06
11:47 a.m.
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