[OpenSIPS-Users] In dialog requests misrouted

Bruce Borrett bruce_borrett at yahoo.com
Tue May 18 23:39:07 CEST 2010


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>
To: OpenSIPS users mailling list <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 
> xx.xxx.0.33;rport=5060;received=41.221.0.33;branch=z9hG4bKb9fc.f1cf4e03.0
> Via: SIP/2.0/UDP xx.xxx.0.42:5060;branch=z9hG4bK3d3a5800;rport=5060
> Record-Route: <sip:xx.xxx.1.13;lr=on;ftag=as5e3b3ce0>
> Record-Route: <sip:xx.xxx.0.33;lr=on;ftag=as5e3b3ce0>
> From: "xxxxxxx7239" <sip:xxxxxxx7239 at xx.xxx.0.42>;tag=as5e3b3ce0
> To: <sip:xxxxxx0114 at xx.xxx.1.13>;tag=as33b0f85f
> Call-ID: 4b3419c86b29e5ce5a387a1b74f7effc at xx.xxx.0.42
> 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:xxxxxxx0114 at xx.xxx.236.105>
> Content-Type: application/sdp
> Content-Length: 290
>
>
> ACK sip:xxxxxx0114 at xx.xxx.1.13:5060 SIP/2.0
> Record-Route: <sip:xx.xxx.0.33;lr=on;ftag=as5e3b3ce0>
> Via: SIP/2.0/UDP xx.xxx.0.33;branch=z9hG4bKb9fc.f1cf4e03.2
> Via: SIP/2.0/UDP xx.xxx.0.42:5060;branch=z9hG4bK5e0350ba;rport=5060
> Route: <sip:xx.xxx.1.13;lr=on;ftag=as5e3b3ce0>
> From: "xxxxxxx7239" <sip:xxxxxxx7239 at xx.xxx.0.42>;tag=as5e3b3ce0
> To: <sip:xxxxxx0114 at xx.xxx.1.13>;tag=as33b0f85f
> Contact: <sip:xxxxxx7239 at xx.xxx.0.42>
> Call-ID: 4b3419c86b29e5ce5a387a1b74f7effc at xx.xxx.0.42
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Max-Forwards: 69
> Content-Length: 0
>
> Thank you very much in advance..
> Bruce
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>  


-- 
Bogdan-Andrei Iancu
www.voice-system.ro


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