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

Bogdan-Andrei Iancu bogdan at voice-system.ro
Sat May 2 09:17:15 CEST 2009 

I think the only thing you can do is (if this path is fixed), to simply 
ignore the Route headers and to do a static routing.

Regards,
Bogdan

Julien Chavanton wrote:
> I think I will try the option to use the "textops" module to enforce 
> the correct order of Record-Route to validate this is my problem etc.
>  
>  
>
> ------------------------------------------------------------------------
> *From:* users-bounces at lists.opensips.org on behalf of Julien Chavanton
> *Sent:* Thu 30/04/2009 3:44 PM
> *To:* Bogdan-Andrei Iancu
> *Cc:* users at lists.opensips.org
> *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route
>
> 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:15141234567 at osip.dev.com>.
> From: "777" <sip:777 at osip.dev.com>;tag=a030735d.
> Call-ID: 8116f933cc4ea03fMjYzN2Q1MGQ5Y2M1ZDc5Yzk4OTRjN2Y5YzEwYWMwMzc..
> CSeq: 1 INVITE.
> Contact: <sip:15141234567 at 2.2.2.2>.
> 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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