20 Mar
2007
20 Mar
'07
9:02 p.m.
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@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@82.215.163.5>;tag=6a4fa6b1
To: <sip:390104491079@82.215.163.67>
Call-ID: 1783604896-30787@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@82.215.133.50 SIP/2.0
Max-Forwards: 69
From: <sip:3405300695@82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A
To: <sip:390104491079@82.215.163.67>
CSeq: 1 INVITE
Content-Type: application/sdp
Call-ID: 11111783604896-30787@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@82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A
To: <sip:390104491079@82.215.163.67>;tag=13FB8534-58
Date: Mon, 25 Mar 2002 04:04:50 GMT
Call-ID: 11111783604896-30787@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@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@82.215.163.5>;tag=6a4fa6b1
To: <sip:390104491079@82.215.163.67>
CSeq: 1 INVITE
Call-ID: 11111783604896-30787@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@82.215.133.50 SIP/2.0
Via: SIP/2.0/UDP 82.215.163.67;branch=z9hG4bK07de.99a696f6.0
From: <sip:3405300695@82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A
Call-ID: 11111783604896-30787@SVIGateway
To: <sip:390104491079@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@82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A
To: <sip:390104491079@82.215.163.67>;tag=13FB8534-58
Date: Mon, 25 Mar 2002 04:04:50 GMT
Call-ID: 11111783604896-30787@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 dont understand nothing :(
Sorry
Good Luck.
Antonio
-----Mensaje original-----
De: tele [mailto:tele@plexia.com]
Enviado el: martes, 20 de marzo de 2007 16:46
Para: users(a)openser.org
CC: antonio.abajo(a)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@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(a)openser.org
[mailto:users-bounces@openser.org] En
nombre
> de tele
> Enviado el: lunes, 19 de marzo de 2007 12:35
> Para: users(a)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(a)openser.org
_______________________________________________
Users mailing list
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Antonio Abajo