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@openser.org [mailto:users-bounces@openser.org] En nombre de tele Enviado el: lunes, 19 de marzo de 2007 12:35 Para: users@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@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
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@openser.org [mailto:users-bounces@openser.org] En nombre de tele Enviado el: lunes, 19 de marzo de 2007 12:35 Para: users@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@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
-
Antonio Abajo -
tele