[OpenSIPS-Users] In dialog requests misrouted
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Tue Jun 15 18:36:47 CEST 2010
Hi Bruce
Bruce Borrett wrote:
> So I believe what you are saying is that it is not optimal to use a
> nat test in a reply route, rather flags should be set in the request
> route and then checked for in the reply route.
that's what I'm doing at least.
> They are using openser 1.3, I have only known openser/opensips since
> 1.5, perhaps flags were not possible in 1.3?
they are :), even in 1.0
> Anyway, they will not change their config for us, so we will need to
> either stick with our hacks, or implement b2bua in between.
yes, that is the bad part when you need to hack your system just to be
able to interconnect with some bogus SIP services :(
Regards,
Bogdan
>
> I will look into opensips b2bua.
>
> Thank you Bogdan.
>
> Regards,
> Bruce
>
> ------------------------------------------------------------------------
> *From:* Bogdan-Andrei Iancu <bogdan at voice-system.ro>
> *To:* OpenSIPS users mailling list <users at lists.opensips.org>
> *Sent:* Thu, 27 May, 2010 15:42:18
> *Subject:* Re: [OpenSIPS-Users] In dialog requests misrouted
>
> Hi Bruce,
>
> by simply looking at a reply, is hard to say if you are the right entity
> to do the fix. Like here, both 1 and 2 see the private contact in the
> reply coming from 4 (same test) but only 2 should do the fixing (it is
> on the NAT edge).
> As best practice, the proxy should remember that the request was send
> somewhere behind a NAT and to accordingly handle the replies (without
> any extra check).
>
> Instead of using asterisk, maybe you can use opensips as B2BUA - it is
> much lighter.
>
> Regards,
> Bogdan
>
> Bruce Borrett wrote:
> > Hi Bogdan
> >
> > I believe I have found the culprit, I managed to get a copy of server
> > 1's config, and they are using uac_nat_test("19") in their reply
> > route, that is completely wrong is it not? Replies would always be
> > fixed? Currently these guys have clients with sbc's connecting
> > straight to their server, and from their server straight to asterisk
> > pstn gateways. We too just had a simple sbc connecting directly to
> > their server. I believe they have hacked there config to a state so
> > fubar, that it can only work in that setup. The only way I think we
> > might solve this without any hacks to our config is to put a bunch of
> > asterisk servers between us and them?
> >
> > Regards,
> > Bruce
> >
> >
> > ------------------------------------------------------------------------
> > *From:* Bogdan-Andrei Iancu <bogdan at voice-system.ro
> <mailto:bogdan at voice-system.ro>>
> > *To:* OpenSIPS users mailling list <users at lists.opensips.org
> <mailto:users at lists.opensips.org>>
> > *Sent:* Tue, 25 May, 2010 21:28:02
> > *Subject:* Re: [OpenSIPS-Users] In dialog requests misrouted
> >
> > Hi Bruce,
> >
> > Bruce Borrett wrote:
> > > Hi Bogdan
> > >
> > > Just to complicate things a bit, 4.4.4.4 is the public ip of the nated
> > > device... I made a mistake in my trace where I changed the contact
> > > address in the initial 200 (from 4.4.4.4), it should have been the
> > > nated ip, 5.5.5.5 for example... So you will see then that 2 is in
> > > fact fixing the contact, but 1 is then fixing it again...
> > why 2 is doing again? if you see a public IP (like 4) in contact, do not
> > fix it again!
> > >
> > > For now we have managed to create a workaround whereby we perform a
> > > "lookup location" on any acks, cancels or byes received from 1.1.1.1,
> > > this function fixes the ruri to the correct one from the location
> > > table, and sends it out correctly. This seems to be working fine so
> > far...
> > yeah, but it is really a ugly hack that my break SIP routing - ACK is a
> > sequential request and must be routed contact + route set only.
> >
> > Regards,
> > Bogdan
> > >
> > > Thank you for your help,
> > >
> > > Regards,
> > > Bruce
> > >
> > >
> > >
> ------------------------------------------------------------------------
> > > *From:* Bogdan-Andrei Iancu <bogdan at voice-system.ro
> <mailto:bogdan at voice-system.ro>
> > <mailto:bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>>>
> > > *To:* OpenSIPS users mailling list <users at lists.opensips.org
> <mailto:users at lists.opensips.org>
> > <mailto:users at lists.opensips.org <mailto:users at lists.opensips.org>>>
> > > *Sent:* Fri, 21 May, 2010 12:57:37
> > > *Subject:* Re: [OpenSIPS-Users] In dialog requests misrouted
> > >
> > > Hi Bruce,
> > >
> > > It is a logical problem. The chain is: 3 -> 1 -> 2 -> 4, and
> when the
> > > reply goes back, the NAT traversal must be done by the border entity
> > > (first in the public net). So, if 4 is behind nat, then 2 must do
> it and
> > > not 1 (like in your case)
> > >
> > > Because 1 (and not 2) is "fixing" the contact , the routing info gets
> > > lost (the 4 hop is lost). So, guilty is 2 for not fixing the
> contact in
> > > 200 OK from 4.
> > >
> > > Regards,
> > > Bogdan
> > >
> > > Bruce Borrett wrote:
> > > > Hi Bogdan
> > > >
> > > > Thank you for your reply, that is exactly how I understood it.
> > > >
> > > > Here is another fuller trace for a better understanding of the
> > > > problem, which I believe to be uac_nat_test. On both servers ,
> when a
> > > > 200 is received, uac_nat_test is returning true because it is
> finding
> > > > the top via (next hop for reply) to be different to the source ip
> > > > (previous hop of reply):
> > > >
> > > > Internet Protocol, Src: 1.1.1.1 (1.1.1.1), Dst: 2.2.2.2 (2.2.2.2)
> > > >
> > > > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
> > > >
> > > > INVITE sip:1111111111 at 2.2.2.2 SIP/2.0
> > > > Record-Route: <sip:1.1.1.1;lr=on;ftag=as1a75bb38>
> > > > Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK5be8.eeb63911.0
> > > > Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK55985828;rport=5060
> > > > From: "22222222222" <sip:2222222222 at 3.3.3.3>;tag=as1a75bb38
> > > > To: <sip:1111111111 at 2.2.2.2>
> > > > Contact: <sip:2222222222 at 3.3.3.3>
> > > > Call-ID: 7a98e4540899dde2053dc8a11cee1a04 at 3.3.3.3
> > > > CSeq: 102 INVITE
> > > > User-Agent: Asterisk PBX
> > > > Max-Forwards: 69
> > > > Date: Tue, 18 May 2010 06:29:00 GMT
> > > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > > > Supported: replaces
> > > > X-CRE-ID: "sbc-tp-04-1274164140.30823.0
> > > > X-DIALSTRING: SIP/ico-bry-001/1111111111
> > > > Content-Type: application/sdp
> > > > Content-Length: 259
> > > >
> > > > v=0
> > > > o=root 4296 4296 IN IP4 3.3.3.3
> > > > s=session
> > > > c=IN IP4 3.3.3.3
> > > > t=0 0
> > > > m=audio 10060 RTP/AVP 18 101
> > > > a=rtpmap:18 G729/8000
> > > > a=fmtp:18 annexb=no
> > > > a=rtpmap:101 telephone-event/8000
> > > > a=fmtp:101 0-16
> > > > a=silenceSupp:off - - - -
> > > > a=ptime:60
> > > > a=sendrecv
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > Internet Protocol, Src: 2.2.2.2 (2.2.2.2), Dst: 4.4.4.4 (4.4.4.4)
> > > >
> > > > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
> > > >
> > > > INVITE sip:1111111111 at 4.4.4.4 SIP/2.0
> > > > Record-Route: <sip:2.2.2.2;lr=on;ftag=as1a75bb38>
> > > > Record-Route: <sip:1.1.1.1;lr=on;ftag=as1a75bb38>
> > > > Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK5be8.91dcf496.0
> > > > Via: SIP/2.0/UDP
> > > > 1.1.1.1;rport=5060;received=1.1.1.1;branch=z9hG4bK5be8.eeb63911.0
> > > > Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK55985828;rport=5060
> > > > From: "2222222222" <sip:2222222222 at 3.3.3.3>;tag=as1a75bb38
> > > > To: <sip:1111111111 at 2.2.2.2>
> > > > Contact: <sip:2222222222 at 3.3.3.3>
> > > > Call-ID: 7a98e4540899dde2053dc8a11cee1a04 at 3.3.3.3
> > > > CSeq: 102 INVITE
> > > > User-Agent: Asterisk PBX
> > > > Max-Forwards: 68
> > > > Date: Tue, 18 May 2010 06:29:00 GMT
> > > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > > > Supported: replaces
> > > > X-CRE-ID: "sbc-tp-04-1274164140.30823.0
> > > > X-DIALSTRING: SIP/ico-bry-001/1111111111
> > > > Content-Type: application/sdp
> > > > Content-Length: 259
> > > >
> > > > v=0
> > > > o=root 4296 4296 IN IP4 3.3.3.3
> > > > s=session
> > > > c=IN IP4 3.3.3.3
> > > > t=0 0
> > > > m=audio 10060 RTP/AVP 18 101
> > > > a=rtpmap:18 G729/8000
> > > > a=fmtp:18 annexb=no
> > > > a=rtpmap:101 telephone-event/8000
> > > > a=fmtp:101 0-16
> > > > a=silenceSupp:off - - - -
> > > > a=ptime:60
> > > > a=sendrecv
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > Internet Protocol, Src: 4.4.4.4 (4.4.4.4), Dst: 2.2.2.2 (2.2.2.2)
> > > >
> > > > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
> > > >
> > > > SIP/2.0 200 OK
> > > > Via: SIP/2.0/UDP
> > 2.2.2.2;branch=z9hG4bK5be8.91dcf496.0;received=2.2.2.2
> > > > Via: SIP/2.0/UDP
> > > > 1.1.1.1;rport=5060;received=1.1.1.1;branch=z9hG4bK5be8.eeb63911.0
> > > > Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK55985828;rport=5060
> > > > Record-Route: <sip:2.2.2.2;lr=on;ftag=as1a75bb38>
> > > > Record-Route: <sip:1.1.1.1;lr=on;ftag=as1a75bb38>
> > > > From: "2222222222" <sip:2222222222 at 3.3.3.3>;tag=as1a75bb38
> > > > To: <sip:1111111111 at 2.2.2.2>;tag=as5bd164c9
> > > > Call-ID: 7a98e4540899dde2053dc8a11cee1a04 at 3.3.3.3
> > > > CSeq: 102 INVITE
> > > > User-Agent: Asterisk PBX 1.6.0.9
> > > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > > > Supported: replaces, timer
> > > > Contact: <sip:1111111111 at 4.4.4.4>
> > > > Content-Type: application/sdp
> > > > Content-Length: 290
> > > >
> > > > v=0
> > > > o=root 2115801714 2115801714 IN IP4 4.4.4.4
> > > > s=Asterisk PBX 1.6.0.9
> > > > c=IN IP4 4.4.4.4
> > > > t=0 0
> > > > m=audio 12004 RTP/AVP 18 101
> > > > a=rtpmap:18 G729/8000
> > > > a=fmtp:18 annexb=no
> > > > a=rtpmap:101 telephone-event/8000
> > > > a=fmtp:101 0-16
> > > > a=silenceSupp:off - - - -
> > > > a=ptime:60
> > > > a=sendrecv
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > Internet Protocol, Src: 2.2.2.2 (2.2.2.2), Dst: 1.1.1.1 (1.1.1.1)
> > > >
> > > > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
> > > >
> > > > SIP/2.0 200 OK
> > > > Via: SIP/2.0/UDP
> > > > 1.1.1.1;rport=5060;received=1.1.1.1;branch=z9hG4bK5be8.eeb63911.0
> > > > Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK55985828;rport=5060
> > > > Record-Route: <sip:2.2.2.2;lr=on;ftag=as1a75bb38>
> > > > Record-Route: <sip:1.1.1.1;lr=on;ftag=as1a75bb38>
> > > > From: "2222222222" <sip:2222222222 at 3.3.3.3>;tag=as1a75bb38
> > > > To: <sip:1111111111 at 2.2.2.2>;tag=as5bd164c9
> > > > Call-ID: 7a98e4540899dde2053dc8a11cee1a04 at 3.3.3.3
> > > > CSeq: 102 INVITE
> > > > User-Agent: Asterisk PBX 1.6.0.9
> > > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > > > Supported: replaces, timer
> > > > Contact: <sip:1111111111 at 4.4.4.4>
> > > > Content-Type: application/sdp
> > > > Content-Length: 290
> > > >
> > > > v=0
> > > > o=root 2115801714 2115801714 IN IP4 4.4.4.4
> > > > s=Asterisk PBX 1.6.0.9
> > > > c=IN IP4 4.4.4.4
> > > > t=0 0
> > > > m=audio 12004 RTP/AVP 18 101
> > > > a=rtpmap:18 G729/8000
> > > > a=fmtp:18 annexb=no
> > > > a=rtpmap:101 telephone-event/8000
> > > > a=fmtp:101 0-16
> > > > a=silenceSupp:off - - - -
> > > > a=ptime:60
> > > > a=sendrecv
> > > >
> > > >
> > > >
> > > >
> > > > ACK is sent with 2.2.2.2 in ruri, instead of 4.4.4.4:
> > > >
> > > > Internet Protocol, Src: 1.1.1.1 (1.1.1.1), Dst: 2.2.2.2 (2.2.2.2)
> > > >
> > > > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
> > > >
> > > > ACK sip:1111111111 at 2.2.2.2:5060 SIP/2.0
> > > > Record-Route: <sip:1.1.1.1;lr=on;ftag=as1a75bb38>
> > > > Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK5be8.eeb63911.2
> > > > Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK07b01eaf;rport=5060
> > > > Route: <sip:2.2.2.2;lr=on;ftag=as1a75bb38>
> > > > From: "2222222222" <sip:2222222222 at 3.3.3.3>;tag=as1a75bb38
> > > > To: <sip:1111111111 at 2.2.2.2>;tag=as5bd164c9
> > > > Contact: <sip:2222222222 at 3.3.3.3>
> > > > Call-ID: 7a98e4540899dde2053dc8a11cee1a04 at 3.3.3.3
> > > > CSeq: 102 ACK
> > > > User-Agent: Asterisk PBX
> > > > Max-Forwards: 69
> > > > Content-Length: 0
> > > >
> > > >
> > > >
> > > >
> > > > The ACK is not relayed on to 4.4.4.4, and so 4.4.4.4 just keeps
> > > > retransmitting 200 replies. Later the BYE from 1.1.1.1 also has an
> > > > incorrect ruri and so it is also not sent on to 4.4.4.4 as follows:
> > > >
> > > > Internet Protocol, Src: 1.1.1.1 (1.1.1.1), Dst: 2.2.2.2 (2.2.2.2)
> > > >
> > > > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
> > > >
> > > > BYE sip:1111111111 at 2.2.2.2:5060 SIP/2.0
> > > > Record-Route: <sip:1.1.1.1;lr=on;ftag=as1a75bb38>
> > > > Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK6be8.2fa52b26.0
> > > > Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK02874f97;rport=5060
> > > > Route: <sip:2.2.2.2;lr=on;ftag=as1a75bb38>
> > > > From: "2222222222" <sip:2222222222 at 3.3.3.3>;tag=as1a75bb38
> > > > To: <sip:1111111111 at 2.2.2.2>;tag=as5bd164c9
> > > > Call-ID: 7a98e4540899dde2053dc8a11cee1a04 at 3.3.3.3
> > > > CSeq: 103 BYE
> > > > User-Agent: Asterisk PBX
> > > > Max-Forwards: 69
> > > > Reason: Q.850 ;cause=16; text="Normal Clearing"
> > > > X-Asterisk-HangupCauseCode: 16
> > > > Content-Length: 0
> > > >
> > > > I believe we would have been able to fix this issue by using the
> > > > dialoq module on both servers, but I do not know much about the
> dialog
> > > > module yet, and unfortunately we have no control over one of the
> > > > opensips servers. i also thought of trying to use the b2bua
> modules on
> > > > just our server, but once again i will first need to learn more
> about
> > > > those, but for now, we have managed to create a workaround
> whereby we
> > > > rewrite the ruri for all acks, byes and cancels with the ip
> retrieved
> > > > from the location table (using avp_db_query), which seems to be
> > > > working, for now. I hope to find a more reliable fix.
> > > >
> > > > Thanks for the help.
> > > > Bruce
> > > >
> > > >
> > > >
> > ------------------------------------------------------------------------
> > > > *From:* Bogdan-Andrei Iancu <bogdan at voice-system.ro
> <mailto:bogdan at voice-system.ro>
> > <mailto:bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>>
> > > <mailto:bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>
> <mailto:bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>>>>
> > > > *To:* OpenSIPS users mailling list <users at lists.opensips.org
> <mailto:users at lists.opensips.org>
> > <mailto:users at lists.opensips.org <mailto:users at lists.opensips.org>>
> > > <mailto:users at lists.opensips.org <mailto:users at lists.opensips.org>
> <mailto:users at lists.opensips.org <mailto:users at lists.opensips.org>>>>
> > > > *Sent:* Tue, 18 May, 2010 17:54:01
> > > > *Subject:* Re: [OpenSIPS-Users] In dialog requests misrouted
> > > >
> > > > Hi Bruce,
> > > >
> > > > The ACK for a 200OK is routed based on the route set - this the RR
> > set +
> > > > the contact of the other party. So, the ACK will have in RURI the
> > > > contact of the other party (from 200 OK) and the RR set as Route
> hdrs.
> > > >
> > > > Regards,
> > > > Bogdan
> > > >
> > > > Bruce Borrett wrote:
> > > > > Hi
> > > > >
> > > > > We are trying to migrate from an SBC to Opensips 1.6. When we are
> > > > > sending calls to another provider who are using Openser, they
> > are not
> > > > > taking the contact address from our 200 replies, instead they are
> > > > > putting our Openser address in the RURI of all Acks, Byes and
> > Cancels.
> > > > > Am I right in saying that this is incorrect? Im not sure where
> they
> > > > > are getting this address either, maybe from the To: field, or
> > from the
> > > > > record route header?
> > > > >
> > > > > Is there a way to match the message received to a transaction and
> > > > > route the message to the contact in the original invite stored
> > by TM?
> > > > > Or perhaps some better way of solving this?
> > > > >
> > > > > Here are the messages:
> > > > >
> > > > > SIP/2.0 200 OK
> > > > > Via: SIP/2.0/UDP
> > > > >
> > > >
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