[OpenSIPS-Users] Multiple response codes being sent

Brad Bendy brad.bendy at benganetworks.com
Tue Aug 24 14:08:45 CEST 2010


Hi Bogdan,

Here is a full trace, breakdown is like this

.2 INVITES to .164
.164 INVITES TO .168
.168 sends a 302 to .164
.164 sends .2 a 503 followed by a 302

.2 should never know about the 302 at all, but it's still getting back
to the originating proxy.

We are not using get_redirects() to do anything with the 302 - from some
Googling and such it appears that might be needed, just not sure how it
would be used.

Thanks for looking at this.

 69.xxx.xxx.2:5060 -> 72.xxx.xxx.164:5060
INVITE sip:6021112222 at 72.xxx.xxx.164 SIP/2.0.
Via: SIP/2.0/UDP 69.xxx.xxx.2:5060;branch=z9hG4bK03578afa;rport.
From: "Test" <sip:500 at 69.xxx.xxx.2>;tag=as4f36ab60.
To: <sip:6021112222 at 72.xxx.xxx.164>.
Contact: <sip:500 at 69.xxx.xxx.2>.
Call-ID: 7b3d78d644ab1f7d52ced54236154da3 at 69.xxx.xxx.2.
CSeq: 102 INVITE.
User-Agent: None.
Max-Forwards: 70.
Remote-Party-ID: "Test" <sip:500 at 69.xxx.xxx.2>;privacy=off;screen=no.
Date: Tue, 24 Aug 2010 12:01:35 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 281.
.
v=0.
o=root 2921 2921 IN IP4 69.xxx.xxx.2.
s=session.
c=IN IP4 69.xxx.xxx.2.
t=0 0.
m=audio 12570 RTP/AVP 18 0 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
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 72.xxx.xxx.164:5060 -> 69.xxx.xxx.2:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 69.xxx.xxx.2:5060;branch=z9hG4bK03578afa;rport=5060.
From: "Test" <sip:500 at 69.xxx.xxx.2>;tag=as4f36ab60.
To: <sip:6021112222 at 72.xxx.xxx.164>.
Call-ID: 7b3d78d644ab1f7d52ced54236154da3 at 69.xxx.xxx.2.
CSeq: 102 INVITE.
Server: OpenSIPS (1.6.2-notls (x86_64/freebsd)).
Content-Length: 0.
.


U 72.xxx.xxx.164:5060 -> 72.xxx.xxx.168:5060
INVITE sip:6021112222 at 72.xxx.xxx.168 SIP/2.0.
Record-Route:
<sip:72.xxx.xxx.164;lr=on;ftag=as4f36ab60;did=36f.3f41d571>.
Via: SIP/2.0/UDP 72.xxx.xxx.164;branch=z9hG4bK23ef.2faf83c4.0.
Via: SIP/2.0/UDP
69.xxx.xxx.2:5060;received=69.xxx.xxx.2;branch=z9hG4bK03578afa;rport=5060.
From: "Test" <sip:500 at 69.xxx.xxx.2>;tag=as4f36ab60.
To: <sip:6021112222 at 72.xxx.xxx.164>.
Contact: <sip:500 at 69.xxx.xxx.2>.
Call-ID: 7b3d78d644ab1f7d52ced54236154da3 at 69.xxx.xxx.2.
CSeq: 102 INVITE.
User-Agent: None.
Max-Forwards: 69.
Remote-Party-ID: "Test" <sip:500 at 69.xxx.xxx.2>;privacy=off;screen=no.
Date: Tue, 24 Aug 2010 12:01:35 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 281.
.
v=0.
o=root 2921 2921 IN IP4 69.xxx.xxx.2.
s=session.
c=IN IP4 69.xxx.xxx.2.
t=0 0.
m=audio 12570 RTP/AVP 18 0 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
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 72.xxx.xxx.168:5060 -> 72.xxx.xxx.164:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 72.xxx.xxx.164;branch=z9hG4bK23ef.2faf83c4.0.
Via: SIP/2.0/UDP
69.xxx.xxx.2:5060;received=69.xxx.xxx.2;branch=z9hG4bK03578afa;rport=5060.
From: "Test" <sip:500 at 69.xxx.xxx.2>;tag=as4f36ab60.
To: <sip:6021112222 at 72.xxx.xxx.164>.
Call-ID: 7b3d78d644ab1f7d52ced54236154da3 at 69.xxx.xxx.2.
CSeq: 102 INVITE.
Server: OpenSIPS (1.6.2-notls (i386/freebsd)).
Content-Length: 0.
.


U 72.xxx.xxx.168:5060 -> 72.xxx.xxx.164:5060
SIP/2.0 302 Moved Temporarily.
Via: SIP/2.0/UDP 72.xxx.xxx.164;branch=z9hG4bK23ef.2faf83c4.0.
Via: SIP/2.0/UDP
69.xxx.xxx.2:5060;received=69.xxx.xxx.2;branch=z9hG4bK03578afa;rport=5060.
From: "Test" <sip:500 at 69.xxx.xxx.2>;tag=as4f36ab60.
To: <sip:6021112222 at 72.xxx.xxx.164>;tag=1235203116.
Call-ID: 7b3d78d644ab1f7d52ced54236154da3 at 69.xxx.xxx.2.
CSeq: 102 INVITE.
Content-Type: application/csv.
Contact: sip:rn=6024810000;npdi;6021112222 at 72.xxx.xxx.168.
User-Agent: eXosip/3.1.0.
Content-Length: 0.
.


U 72.xxx.xxx.164:5060 -> 72.xxx.xxx.168:5060
ACK sip:6021112222 at 72.xxx.xxx.168 SIP/2.0.
Via: SIP/2.0/UDP 72.xxx.xxx.164;branch=z9hG4bK23ef.2faf83c4.0.
From: "Test" <sip:500 at 69.xxx.xxx.2>;tag=as4f36ab60.
Call-ID: 7b3d78d644ab1f7d52ced54236154da3 at 69.xxx.xxx.2.
To: <sip:6021112222 at 72.xxx.xxx.164>;tag=1235203116.
CSeq: 102 ACK.
Max-Forwards: 70.
User-Agent: OpenSIPS (1.6.2-notls (x86_64/freebsd)).
Content-Length: 0.
.


U 72.xxx.xxx.164:5060 -> 69.xxx.xxx.2:5060
SIP/2.0 503 No more routes
Via: SIP/2.0/UDP 69.xxx.xxx.2:5060;branch=z9hG4bK03578afa;rport=5060.
From: "Test" <sip:500 at 69.xxx.xxx.2>;tag=as4f36ab60.
To:
<sip:6021112222 at 72.xxx.xxx.164>;tag=f254695ad980185f5ba46cc313375d56.4b85.
Call-ID: 7b3d78d644ab1f7d52ced54236154da3 at 69.xxx.xxx.2.
CSeq: 102 INVITE.
Server: OpenSIPS (1.6.2-notls (x86_64/freebsd)).
Content-Length: 0.
.


U 72.xxx.xxx.164:5060 -> 69.xxx.xxx.2:5060
SIP/2.0 302 Moved Temporarily.
Via: SIP/2.0/UDP
69.xxx.xxx.2:5060;received=69.xxx.xxx.2;branch=z9hG4bK03578afa;rport=5060.
From: "Test" <sip:500 at 69.xxx.xxx.2>;tag=as4f36ab60.
To: <sip:6021112222 at 72.xxx.xxx.164>;tag=1235203116.
Call-ID: 7b3d78d644ab1f7d52ced54236154da3 at 69.xxx.xxx.2.
CSeq: 102 INVITE.
Content-Type: application/csv.
Contact: sip:rn=6024810000;npdi;6021112222 at 72.xxx.xxx.168.
User-Agent: eXosip/3.1.0.
Content-Length: 0.
.


U 69.xxx.xxx.2:5060 -> 72.xxx.xxx.164:5060
ACK sip:6021112222 at 72.xxx.xxx.164 SIP/2.0.
Via: SIP/2.0/UDP 69.xxx.xxx.2:5060;branch=z9hG4bK03578afa;rport.
From: "Test" <sip:500 at 69.xxx.xxx.2>;tag=as4f36ab60.
To:
<sip:6021112222 at 72.xxx.xxx.164>;tag=f254695ad980185f5ba46cc313375d56.4b85.
Contact: <sip:500 at 69.xxx.xxx.2>.
Call-ID: 7b3d78d644ab1f7d52ced54236154da3 at 69.xxx.xxx.2.
CSeq: 102 ACK.
User-Agent: None.
Max-Forwards: 70.
Remote-Party-ID: "Test" <sip:500 at 69.xxx.xxx.2>;privacy=off;screen=no.
Content-Length: 0.
.


U 69.xxx.xxx.2:5060 -> 72.xxx.xxx.164:5060
ACK sip:6021112222 at 72.xxx.xxx.164 SIP/2.0.
Via: SIP/2.0/UDP 69.xxx.xxx.2:5060;branch=z9hG4bK03578afa;rport.
From: "Test" <sip:500 at 69.xxx.xxx.2>;tag=as4f36ab60.
To: <sip:6021112222 at 72.xxx.xxx.164>;tag=1235203116.
Contact: <sip:500 at 69.xxx.xxx.2>.
Call-ID: 7b3d78d644ab1f7d52ced54236154da3 at 69.xxx.xxx.2.
CSeq: 102 ACK.
User-Agent: None.
Max-Forwards: 70.
Remote-Party-ID: "Test" <sip:500 at 69.xxx.xxx.2>;privacy=off;screen=no.
Content-Length: 0.

On Tue, 2010-08-24 at 10:57 +0300, Bogdan-Andrei Iancu wrote:

> Hi Brad,
> 
> Maybe I do not fully understand your case, but opensips is not sending a 
> 302 after 200 OK...Maybe you can post the call flow (a SIP trace) from 
> the SIP server showing the entire scenario.
> 
> Regards,
> Bogdan
> 
> Brad Bendy wrote:
> > Hi,
> >
> > Im having a heck of a time figuring this out:
> >
> > INVITE comes to our switch, we send a INVITE to another proxy that 
> > responds with a 302, we parse that 302 in failure route then use a 
> > route() command to go to another route block which does some other 
> > processing (will send out more INVITE's, do certain things on failure, 
> > etc), if the original call does get canceled or completes successfully 
> > with a 200 OK the originating proxy receives the original 302 request 
> > plus what ever our final failure response code we want to send.
> >
> > The behavior does seem correct as openSIPs is just forwarding  the 
> > 302, but in this case I want it to send only the final response code 
> > back to the originating client.
> >
> > The initital route block which sends the INVITE to get the 302 is very 
> > simple, we just write the rU and rd and send via t_relay, 
> > onreply_route does a little parsing then failure_route sends to a new 
> > block.
> >
> > Any help on this would be great, I think it's my logic in the switch 
> > that is wrong somewhere.
> >
> > Thanks!
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >   
> 
> 

-- 
Brad Bendy
Chief Technical Officer
brad.bendy at benganetworks.com

Benga Networks, LLC.
10115 E. Bell Rd, Ste. 107-451
Scottsdale, AZ 85260-2189

Toll Free:    877-44-BENGA
Local:          480-970-5200
Cell:             602-550-4004
Fax:             866-852-4468
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