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

Julien Chavanton jc at atlastelecom.com
Thu Apr 30 13:40:21 CEST 2009


 
 
UA --> PROXY 1.1.1.1 --> PROXY 2.2.2.2 --> UA
 
P1 --> P2
INVITE 
Record-Route: <sip:1.1.1.1;lr=on;nat=yes>
 
P2 --> P1
100 Trying
Record-Route: <sip:1.1.1.1;lr=on;nat=yes>
Record-Route: <sip:2.2.2.2:5060;lr>
 
 
Is there something wrong ? shouldn't proxy 2.2.2.2 add his Record-Route on top of the existing Record-Route ?

________________________________

From: Bogdan-Andrei Iancu [mailto:bogdan at voice-system.ro]
Sent: Thu 30/04/2009 8:12 AM
To: Julien Chavanton
Cc: users at lists.opensips.org
Subject: Re: [OpenSIPS-Users] handling multiple proxy / Record-Route



Hi Julien,

I think Asterisk is doing the job properly. As you see the 200 OK has:
    Contact: <sip:15141234567 at 2.2.2.2:5060>.
    Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.
  Record-Route: <sip:2.2.2.2:5060;lr>.

So, Asterisk is generating the ACK with the Contact in RURI and the
Route set in the reverted order (correct loose routing).
    -> RURI: sip:15141234567 at 2.2.2.2:5060
           Destination: sip:2.2.2.2:5060;lr
     Route: sip:2.2.2.2:5060;lr + sip:1.1.1.1;lr=on;nat=yes

I think the problem here is who and why adding the bottom RR in 200 OK
(why 2 of them ?)

Regards,
Bogdan

Julien Chavanton wrote:
>
> 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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