[OpenSIPS-Users] handling multiple proxy / Record-Route

Wed Apr 29 21:44:49 CEST 2009

Hi,

I have a situation whit multiple proxy where ACK is not sent as I would expect. 

if we look at the following "200 OK", I am expecting ACK to be sent to 1.1.1.1 but the "Asterisk PBX 1.6.0.6." is selecting 2.2.2.2 is this normal ?

Do I have to handle Record-Route differently ?

U 1.1.1.1:5060 -> 192.168.1.108:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
To: <sip:15141234567 at osip.dev.com>;tag=as664de2c2.
From: "15141234567" <sip:15141234567 at 192.168.1.108>;tag=as55bd7355.
Call-ID: 641cab3f73fa37a871818d1a70c4061b at 192.168.1.108 <mailto:641cab3f73fa37a871818d1a70c4061b at 192.168.1.108> .
CSeq: 102 INVITE.
Content-Type: application/sdp.
Contact: <sip:15141234567 at 2.2.2.2:5060>.
Content-Length: 241.
Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
User-Agent: Packetrino.
Supported: replaces.
Record-Route: <sip:2.2.2.2:5060;lr>.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.

---------------------------------------------------------

complete SIP signaling

---------------------------------------------------------

#
U 192.168.1.108:5060 -> 1.1.1.1:5060
INVITE sip:15141234567 at osip.dev.com SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport.
Max-Forwards: 70.
From: "15141234567" <sip:15141234567 at 192.168.1.108>;tag=as55bd7355.
To: <sip:15141234567 at osip.dev.com>.
Contact: <sip:15141234567 at 192.168.1.108>.
Call-ID: 641cab3f73fa37a871818d1a70c4061b at 192.168.1.108 <mailto:641cab3f73fa37a871818d1a70c4061b at 192.168.1.108> .
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 1.6.0.6.
Date: Wed, 29 Apr 2009 15:38:18 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 265.
.
v=0.
o=root 1992389746 1992389746 IN IP4 192.168.1.108.
s=Asterisk PBX 1.6.0.6.
c=IN IP4 192.168.1.108.
t=0 0.
m=audio 11232 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

#
U 1.1.1.1:5060 -> 192.168.1.108:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport=5060;received=74.56.45.88.
From: "15141234567" <sip:15141234567 at 192.168.1.108>;tag=as55bd7355.
To: <sip:15141234567 at osip.dev.com>.
Call-ID: 641cab3f73fa37a871818d1a70c4061b at 192.168.1.108 <mailto:641cab3f73fa37a871818d1a70c4061b at 192.168.1.108> .
CSeq: 102 INVITE.
Server: OpenSIPS (1.4.4-notls (x86_64/linux)).
Content-Length: 0.
.

#
U 1.1.1.1:5060 -> 192.168.1.108:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
To: <sip:15141234567 at osip.dev.com>;tag=as664de2c2.
From: "15141234567" <sip:15141234567 at 192.168.1.108>;tag=as55bd7355.
Call-ID: 641cab3f73fa37a871818d1a70c4061b at 192.168.1.108 <mailto:641cab3f73fa37a871818d1a70c4061b at 192.168.1.108> .
CSeq: 102 INVITE.
Content-Type: application/sdp.
Contact: <sip:15141234567 at 2.2.2.2:5060>.
Content-Length: 241.
Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
User-Agent: Packetrino.
Supported: replaces.
Record-Route: <sip:2.2.2.2:5060;lr>.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
.
v=0.
o=root 29378 29378 IN IP4 64.2.142.160.
s=session.
c=IN IP4 1.1.1.1.
t=0 0.
m=audio 52528 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

#
U 1.1.1.1:5060 -> 192.168.1.108:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
To: <sip:15141234567 at osip.dev.com>;tag=as664de2c2.
From: "15141234567" <sip:15141234567 at 192.168.1.108>;tag=as55bd7355.
Call-ID: 641cab3f73fa37a871818d1a70c4061b at 192.168.1.108 <mailto:641cab3f73fa37a871818d1a70c4061b at 192.168.1.108> .
CSeq: 102 INVITE.
Contact: <sip:15141234567 at 2.2.2.2:5060>.
Content-Length: 0.
Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
User-Agent: Packetrino.
Supported: replaces.
Record-Route: <sip:2.2.2.2:5060;lr>.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
.

#
U 1.1.1.1:5060 -> 192.168.1.108:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.
To: <sip:15141234567 at osip.dev.com>;tag=as664de2c2.
From: "15141234567" <sip:15141234567 at 192.168.1.108>;tag=as55bd7355.
Call-ID: 641cab3f73fa37a871818d1a70c4061b at 192.168.1.108 <mailto:641cab3f73fa37a871818d1a70c4061b at 192.168.1.108> .
CSeq: 102 INVITE.
Content-Type: application/sdp.
Contact: <sip:15141234567 at 2.2.2.2:5060>.
Content-Length: 241.
Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
User-Agent: Packetrino.
Supported: replaces.
Record-Route: <sip:2.2.2.2:5060;lr>.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
.
v=0.
o=root 29378 29379 IN IP4 64.2.142.160.
s=session.
c=IN IP4 1.1.1.1.
t=0 0.
m=audio 52528 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

#
U 192.168.1.108:5060 -> 2.2.2.2:5060
ACK sip:15141234567 at 2.2.2.2:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK04335252;rport.
Route: <sip:2.2.2.2:5060;lr>,<sip:1.1.1.1;lr=on;nat=yes>.
Max-Forwards: 70.
From: "15141234567" <sip:15141234567 at 192.168.1.108>;tag=as55bd7355.
To: <sip:15141234567 at osip.dev.com>;tag=as664de2c2.
Contact: <sip:15141234567 at 192.168.1.108>.
Call-ID: 641cab3f73fa37a871818d1a70c4061b at 192.168.1.108 <mailto:641cab3f73fa37a871818d1a70c4061b at 192.168.1.108> .
CSeq: 102 ACK.
User-Agent: Asterisk PBX 1.6.0.6.
Content-Length: 0.
.

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