RV: RV: [Users] WeSIP session question
Antonio Abajo
antonio.abajo at voztele.com
Tue Mar 20 19:02:22 CET 2007
Buenas tardes Tele
In the capture, I see the following call flow.
GW WESIP MS
------ INV(S1)-----> -------INV (S2) ------>
<--------183 -----------
------- 183 ----------->
<----------ACK ---------
<---------503 ----------
...
14:07:57 19Mar2007 DEBUG SipProcessor [SipProcessor[4]]- <<<<<<<<< Request
Received <<<<<<<<<
INVITE sip:390104491079 at 82.215.163.67 SIP/2.0
Via: SIP/2.0/UDP 82.215.163.5:5060;branch=z9hG4bK6a4fa6b1
Max-Forwards: 69
From: <sip:3405300695 at 82.215.163.5>;tag=6a4fa6b1
To: <sip:390104491079 at 82.215.163.67>
Call-ID: 1783604896-30787 at SVIGateway
CSeq: 1 INVITE
Contact: <sip:82.215.163.5:5060>
Content-Type: application/sdp
Content-Length: 213
14:07:58 19Mar2007 DEBUG SipRequest [SipProcessor[4]]- >>>>>>>>> Sending
Request >>>>>>>>>
INVITE sip:199 at 82.215.133.50 SIP/2.0
Max-Forwards: 69
From: <sip:3405300695 at 82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A
To: <sip:390104491079 at 82.215.163.67>
CSeq: 1 INVITE
Content-Type: application/sdp
Call-ID: 11111783604896-30787 at SVIGateway
Contact: <sip:82.215.163.67:5060;transport=udp>
Via: SIP/2.0/UDP 82.215.163.67:5060;branch=z9hG4bK477735439
Content-Length: 21314:07:58 19Mar2007 DEBUG SipProcessor [SipProcessor[4]]-
<<<<<<<<< Response Received <<<<<<<<<
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 82.215.163.67:5060;branch=z9hG4bK477735439
From: <sip:3405300695 at 82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A
To: <sip:390104491079 at 82.215.163.67>;tag=13FB8534-58
Date: Mon, 25 Mar 2002 04:04:50 GMT
Call-ID: 11111783604896-30787 at SVIGateway
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 INVITE
Allow:
INVITE,CANCEL,ACK,BYE,INFO,OPTIONS,UPDATE,REGISTER,SUBSCRIBE,NOTIFY,PRACK,RE
FER
Allow-Events: telephone-event
Contact: <sip:199 at 82.215.133.50:5060>
Content-Type: application/sdp
Content-Disposition: oSystemsSIP-GW-UserAg
Content-Length: 235
14:07:58 19Mar2007 DEBUG SipResponse [SipProcessor[4]]- >>>>>>>>> Sending
Response >>>>>>>>>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 82.215.163.5:5060;branch=z9hG4bK6a4fa6b1
Max-Forwards: 69
From: <sip:3405300695 at 82.215.163.5>;tag=6a4fa6b1
To: <sip:390104491079 at 82.215.163.67>
CSeq: 1 INVITE
Call-ID: 11111783604896-30787 at SVIGateway
Content-Type: application/sdp
Content-Length: 235
v=0
o=CiscoSystemsSIP-GW-UserAgent 9756 1908 IN IP4 82.215.133.50
s=SIP Call
c=IN IP4 82.215.133.50
t=0 0
m=audio 19450 RTP/AVP 0 99
c=IN IP4 82.215.133.50
a=rtpmap:0 PCMU/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
14:08:07 19Mar2007 DEBUG SipProcessor [SipProcessor[3]]- <<<<<<<<< Request
Received <<<<<<<<<
ACK sip:199 at 82.215.133.50 SIP/2.0
Via: SIP/2.0/UDP 82.215.163.67;branch=z9hG4bK07de.99a696f6.0
From: <sip:3405300695 at 82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A
Call-ID: 11111783604896-30787 at SVIGateway
To: <sip:390104491079 at 82.215.163.67>;tag=13FB8534-58
CSeq: 1 ACK
User-Agent: OpenSer (1.2.0-pre6-notls (i386/linux))
Content-Length: 0
14:08:07 19Mar2007 DEBUG SipProcessor [SipProcessor[4]]- <<<<<<<<< Response
Received <<<<<<<<<
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 82.215.163.67:5060;branch=z9hG4bK477735439
From: <sip:3405300695 at 82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A
To: <sip:390104491079 at 82.215.163.67>;tag=13FB8534-58
Date: Mon, 25 Mar 2002 04:04:50 GMT
Call-ID: 11111783604896-30787 at SVIGateway
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=41
Content-Length: 0
The signalling is not correct but, I have looked your code and believe that
it is right, revise it and you check if you have the last version of WeSIP
and you return to try, because the last release commited in HYPERLINK
"http://www.wesip.com/mediawiki/index.php/Downloads:WeSIP_Beta_0.1"http://ww
w.wesip.com/mediawiki/index.php/Downloads:WeSIP_Beta_0.1 was fixed a bug
with the ApplicationSessions.
In my test application I reply your behaviour(with 5xx, with AppSession
attributes) and all have working perfectly, I don’t understand nothing :(
Sorry
Good Luck.
Antonio
-----Mensaje original-----
De: tele [mailto:tele at plexia.com]
Enviado el: martes, 20 de marzo de 2007 16:46
Para: users at openser.org
CC: antonio.abajo at voztele.com
Asunto: Re: RV: [Users] WeSIP session question
Hola Antonio,
Your example works great and the code is more clear and correct than
mine :)
i was not clear, my problem was only with the 503 Service Unavailable
response, that means a 5xx Server Failure, is it possibile that when the
container or openser with seas recive a 5xx response is not able to do
for example a creation of a new invite to UA3 recover from the same
session?. Or probably i'm occur in a 473 response for an incorrect use
of SipApplicationSession.
anyway with a 480 Temporarily Unavailable response from the UA2 i solve
my problem.
thank you very much
:tele
On Mon, 2007-03-19 at 19:27 +0100, Antonio Abajo wrote:
> Hola tele ;)
>
> I have tested an example similar to yours, and it works ok. It's de
> following:
>
> UA1 ---[1]--> WeSIP --[2]--486 Busy Here---> UA2
> |
> [3]
> ---OK----_> UA3
>
> [1] UA1 generates an INVITE to UA2 ant it 's forwarded to UA2 with a B2BUA
> behaviour. In do INVITE. I have implemented the following:
>
> protected void doInvite(SipServletRequest invite){
> SipServletRequest otherInvite = sf.createRequest(invite, false);
> SipURI sipUri = sf.createSipURI("UA2","proxy");
> otherInvite.setRequestURI(sipUri);
> otherInvite.getSession().setAttribute("REQUEST", otherInvite);
>
> otherInvite.getSession().setAttribute("PEER_SESSION",invite.getSession());
>
> invite.getSession().setAttribute("PEER_SESSION",otherInvite.getSession());
> invite.getSession().setAttribute("REQUEST", invite);
> otherInvite.send();
> }
>
> [2] UA2 declines the call and a 486 response is received by WeSIP and
> process it in doErrorResponse:
>
> protected void doErrorResponse(SipServletResponse errorResponse) {
> switch(errorResponse.getStatus()){
> case 486:
> SipServletRequest request = (SipServletRequest)
> errorResponse.getSession().getAttribute("REQUEST");
> SipSession otherSession = (SipSession)
> errorResponse.getSession().getAttribute("PEER_SESSION");
> SipURI sipUri = sf.createSipURI("UA3", "proxy");
> request.setRequestURI(sipUri);
> SipServletRequest newRequest =
> sf.createRequest(request,false);
> newRequest.getSession().setAttribute("REQUEST", newRequest);
> newRequest.getSession().setAttribute("PEER_SESSION",
> otherSession);
> otherSession.setAttribute("PEER_SESSION",
> newRequest.getSession());
> newRequest.setHeader("X-SSVTPBX", "yes");
> newRequest.send();
> break;
> }
> }
>
> [3] UA3 Recevies the second call and it takes down.
>
>
> I do not understand the cause of error. I annex you the code of simple sip
> application example that replies to your problem. You can start it up and
> checking if it works. Else if work for you, I would attempt watching the
> particular case when a 5XX responses are received.
>
> Sorry...
>
> Best regards.
> Antonio.
>
> -----Mensaje original-----
> De: tele [mailto:tele at plexia.com]
> Enviado el: lunes, 19 de marzo de 2007 14:33
> Para: Antonio Abajo Álvarez
> Asunto: RE: [Users] WeSIP session question
>
> Hola Antonio,
>
> The scenario is more complex, i'll try to explain it:
>
> PSTN
> |
> |
> UA---->GW ------> WeSIP(B2BUA)
> |
> |
> mediaserver
>
>
> GW: 82.215.163.5
> WeSIP: 82.215.163.67
> MS: 82.215.133.50
>
> in the mediaserver there is an vxml script that i play in early media
> and in case of particular event return to wesip a 503 temporaly
> unavailable or e 410 Gone. So my B2BUA application have control of this
> and can do stuff with the 503 and 410.
>
> in particular, in case of 503 temporaly unavailable i get the upstream
> session and create a new invite to the media server for play another
> announcement associated. in case of 410 gone i generate a new invite to
> the GW with the original URI request for the correct termination.
>
> Yes when the session is removed i'm able to send another call.
>
> attached here the logs and the servlet.
>
> don't care about the hardcoded IP and the repeated code :-)
> i'm doing testing..
>
> regards,
>
> :tele
>
>
> On Mon, 2007-03-19 at 13:29 +0100, Antonio Abajo Álvarez wrote:
> > Hi Tele,
> >
> > I don't understand very well the problem., for what I understand you
have
> > the following:
> >
> >
> > UA1 --------------------- WeSIP --------------------- UA2
> >
> > ---INV/4XX-6XX/ACK (SS1) --> ----INV/4XX-6XX/ACK(SS2)-->
> >
> > You try to send another call and receive the 473 response of WeSIP...
> >
> > ------INV/473/ACK (SS1) -->
> >
> > And when the session has been removed you can send another call.
> >
> > If it is the case, I understand that the 473 response is send from
> > application or from openser script configuration, because the internal
> > behaviour of SIP doesn't send this response automatically.
> >
> > Can you verify if it is the case?
> >
> > Thank you very much...
> >
> > Antonio.
> >
> > -----Mensaje original-----
> > De: users-bounces at openser.org [mailto:users-bounces at openser.org] En
nombre
> > de tele
> > Enviado el: lunes, 19 de marzo de 2007 12:35
> > Para: users at openser.org
> > Asunto: [Users] WeSIP session question
> >
> > Hi,
> >
> > I've a problem with WeSIP in B2BUA mode, in case of failed call 4xx-6xx
> > correctly terminated, when i try to send another call to WeSIP i recive
> > a "473 Filtered destination" then for send another call i've to wait
> > WeSIP complete some management with session:
> >
> > 14:10:16 19Mar2007 DEBUG SipConnector [SipProcessor[3]]- recycle:
> > Recycling processor SipProcessor[3]
> > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager
> > [StandardAppSessionManager[/inapp]]- AppSession Id
> > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =1
> > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager
> > [StandardAppSessionManager[/inapp]]- SipSession
> > [z9hG4bK69e6006f] in state [3] with lifetime of :74490
> > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager
> > [StandardAppSessionManager[/inapp]]- AppSession Id
> > [E3E1D9414BC47572CBC73EC7B4A53531] with a number of sessions =1
> > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager
> > [StandardAppSessionManager[/inapp]]- SipSession
> > [z9hG4bK69e68616] in state [3] with lifetime of :40281
> > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager
> > [StandardAppSessionManager[/inapp]]- AppSession Id
> > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =1
> > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager
> > [StandardAppSessionManager[/inapp]]- SipSession
> > [z9hG4bK69e6006f] in state [3] with lifetime of :134500
> > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager
> > [StandardAppSessionManager[/inapp]]- AppSession Id
> > [E3E1D9414BC47572CBC73EC7B4A53531] with a number of sessions =1
> > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager
> > [StandardAppSessionManager[/inapp]]- SipSession
> > [z9hG4bK69e68616] in state [3] with lifetime of :100291
> > 14:12:56 19Mar2007 DEBUG StandardAppSessionManager
> > [StandardAppSessionManager[/inapp]]- AppSession Id
> > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =0
> > 14:12:56 19Mar2007 DEBUG StandardAppSessionManager
> > [StandardAppSessionManager[/inapp]]- Remove AppSession
> > [B4D6C9C4288784A68E032D20AB78BD8E]
> >
> > When i see Remove AppSession i'm able to send another call...
> > I've read the sip servlet spec about that and i'm trying to invalidate()
> > or setExpires() to SiApplicationSession in case of failed call. but it's
> > not clear how to do yet.
> >
> > i can provide the full debug if needed.
> >
> > thank you!
> >
> > regards
> >
> > :tele
> >
> >
> >
> > _______________________________________________
> > Users mailing list
> > Users at openser.org
> > http://openser.org/cgi-bin/mailman/listinfo/users
>
> _______________________________________________
> Users mailing list
> Users at openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users
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