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

Bogdan-Andrei Iancu bogdan at voice-system.ro
Sat May 2 09:15:35 CEST 2009 

Hi Julien,

as the bogus proxy is the last on the path (just before the client), it 
is not much you can do about.

Even if you try to fix the order in the 200 OK reply, it will not work 
(only partial) as the callee will still have the bogus order, so it will 
not be able to route to the caller.

Regards,
Bogdan

Julien Chavanton wrote:
> thank you, this is a problem as I do not control this proxy (2.2.2.2), 
> is there a suggested way of handling this problem ?
>  
> Maybe there is something esle wrong on my side cusaing this problem so 
> I am including the SIP communication between the proxy this time
>  
>  
>  
> #
> U 1.1.1.1:5060 -> 2.2.2.2:5060
> INVITE sip:15148622633 at 2.2.2.2 SIP/2.0.
> Record-Route: <sip:1.1.1.1;lr>.
> Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK09e6.36a0f975.0.
> Via: SIP/2.0/UDP 
> 10.0.1.74:58366;received=10.0.1.74;branch=z9hG4bK-d87543-0f348609f47bda44-1--d87543-;rport=58366.
> Max-Forwards: 69.
> Contact: <sip:777 at 10.0.1.74:58366>.
> To: "15141234567"<sip:15148622633 at osip.dev.com>.
> From: "777"<sip:777 at osip.dev.com>;tag=a030735d.
> Call-ID: 8116f933cc4ea03fMjYzN2Q1MGQ5Y2M1ZDc5Yzk4OTRjN2Y5YzEwYWMwMzc..
> CSeq: 1 INVITE.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, 
> SUBSCRIBE, INFO.
> Content-Type: application/sdp.
> User-Agent: eyeBeam release 1003s stamp 31159.
> Content-Length: 478.
> P-hint: Route[6]: mediaproxy .
> .
> v=0.
> o=- 8 2 IN IP4 10.0.1.74.
> s=CounterPath eyeBeam 1.5.
> c=IN IP4 1.1.1.1.
> t=0 0.
> m=audio 52550 RTP/AVP 0 8 18 101.
> a=alt:1 4 : LM6OZaAl 4x8r9qea 192.168.1.101 50006.
> a=alt:2 3 : 84SVypDj oi4PbxZ7 192.168.114.1 50006.
> a=alt:3 2 : L4wf6+MH s4gK5GAV 192.168.146.1 50006.
> a=alt:4 1 : cg2pbkCG WDFvj29+ 10.0.1.74 50006.
> a=fmtp:18 annexb=no.
> a=fmtp:101 0-15.
> a=rtpmap:101 telephone-event/8000.
> a=sendrecv.
> a=x-rtp-session-id:D56BCBC26473491FA111854E4C9F3575.
> a=direction:active.
> #
> U 2.2.2.2:5060 -> 1.1.1.1:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP 
> 1.1.1.1:5060;branch=z9hG4bK09e6.36a0f975.0;received=1.1.1.1;rport=5060.
> Via: SIP/2.0/UDP 
> 10.0.1.74:58366;received=10.0.1.74;branch=z9hG4bK-d87543-0f348609f47bda44-1--d87543-;rport=58366.
> To: "15141234567" <sip:15148622633 at osip.dev.com>.
> From: "777" <sip:777 at osip.dev.com>;tag=a030735d.
> Call-ID: 8116f933cc4ea03fMjYzN2Q1MGQ5Y2M1ZDc5Yzk4OTRjN2Y5YzEwYWMwMzc..
> CSeq: 1 INVITE.
> Contact: <sip:15148622633 at 64.2.142.75>.
> Content-Length: 0.
> Record-Route: <sip:1.1.1.1;lr>.
> User-Agent: Packetrino.
> Supported: replaces.
> Record-Route: <sip:2.2.2.2:5060;lr>.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> .
>  
>
> ------------------------------------------------------------------------
> *From:* Bogdan-Andrei Iancu [mailto:bogdan at voice-system.ro]
> *Sent:* Thu 30/04/2009 3:44 PM
> *To:* Julien Chavanton
> *Cc:* users at lists.opensips.org
> *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route
>
> 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.
> > > .
> > >
> > >
> > > 
> ------------------------------------------------------------------------
> > >
> > > _______________________________________________
> > > Users mailing list
> > > Users at lists.opensips.org
> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > > 
> >
>

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