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

Thu Apr 30 16:44:48 CEST 2009

Hi Julian,

Julien Chavanton wrote:
>  
>  
> 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>
>  
^^^^^^^^^^^^

This is not correct. The RR of P2 most me on top of RR of P1 - adding RR 
headers works as a stack.

Regards,
Bogdan
>  
> 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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