Hi,
Who is inchage of replacing the to header in the message (openser/wesip)? When wesip get the message the to uri is still with the openser IP. Do I need to do something when the request is INVITE request or these openser replace it automatically before he send it to wesip?
Please advice. I realy need it soon. Thanks in advance, Shie
El Sábado, 15 de Marzo de 2008, shiebar@post.tau.ac.il escribió:
Hi,
Who is inchage of replacing the to header in the message (openser/wesip)? When wesip get the message the to uri is still with the openser IP. Do I need to do something when the request is INVITE request or these openser replace it automatically before he send it to wesip?
Please advice. I realy need it soon. Thanks in advance,
99% you don't need at all to change the "To" header. That's RFC 3261 non complinat.
Hi Inaki,
Then what should I do if I get the following error: SIP/2.0 481 Call Leg/Transaction Does Not Exist
When both caller and callee are online? WeSIP receive the request with the real From uri but the To uri contains the IP of openser server. Is this the correct situation?
Thanks & regards, Shie
Quoting Iñaki Baz Castillo ibc@aliax.net:
El Sábado, 15 de Marzo de 2008, shiebar@post.tau.ac.il escribió:
Hi,
Who is inchage of replacing the to header in the message (openser/wesip)? When wesip get the message the to uri is still with the openser IP. Do I need to do something when the request is INVITE request or these openser replace it automatically before he send it to wesip?
Please advice. I realy need it soon. Thanks in advance,
99% you don't need at all to change the "To" header. That's RFC 3261 non complinat.
-- Iñaki Baz Castillo
Users mailing list Users@lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users
El Sábado, 15 de Marzo de 2008, shiebar@post.tau.ac.il escribió:
Hi Inaki,
Then what should I do if I get the following error: SIP/2.0 481 Call Leg/Transaction Does Not Exist
When both caller and callee are online?
And why do you assume that it's a fail related to the "To" header?
RFC 3261 21.4.19 481 Call/Transaction Does Not Exist
This status indicates that the UAS received a request that does not match any existing dialog or transaction.
Also read how a UA tries to match an incoming request to an existing transaction: http://tools.ietf.org/html/rfc3261#section-17.2.3
WeSIP receive the request with the real From uri but the To uri contains the IP of openser server. Is this the correct situation?
Sincerely, you should read the RFC 3261 to understand that the "To" header is usually not examinated by the UAS, just the Request-Uri. But you cannot asumme something incorrect without reading about it and later ask "then what?".
;)
Hi Inaki,
I have already read it but probably missed this important detail. Thanks for the advice. I will read it again more carefully.
Best regards, Shie
Quoting Iñaki Baz Castillo ibc@aliax.net:
El Sábado, 15 de Marzo de 2008, shiebar@post.tau.ac.il escribió:
Hi Inaki,
Then what should I do if I get the following error: SIP/2.0 481 Call Leg/Transaction Does Not Exist
When both caller and callee are online?
And why do you assume that it's a fail related to the "To" header?
RFC 3261 21.4.19 481 Call/Transaction Does Not Exist
This status indicates that the UAS received a request that does not match any existing dialog or transaction.
Also read how a UA tries to match an incoming request to an existing transaction: http://tools.ietf.org/html/rfc3261#section-17.2.3
WeSIP receive the request with the real From uri but the To uri contains the IP of openser server. Is this the correct situation?
Sincerely, you should read the RFC 3261 to understand that the "To" header is usually not examinated by the UAS, just the Request-Uri. But you cannot asumme something incorrect without reading about it and later ask "then what?".
;)
-- Iñaki Baz Castillo
Users mailing list Users@lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users
El Sábado, 15 de Marzo de 2008, shiebar@post.tau.ac.il escribió:
Hi Inaki,
I have already read it but probably missed this important detail. Thanks for the advice. I will read it again more carefully.
In resume: The real destination is the Request-Uri (except if there is a "Route" header). "To" header jist means the **original** destination. For example:
- A calls B so it generates an INVITE like: INVITE B@domain1 SIP/2.0 To: B@domain1 From: A@domain1 (note Request-Uri = "To" header)
- A sends the INVITE to the proxy responsible for domain1.
- The proxy has has a diversion to C@domain2 for B.
- Proxy rewrites the request-Uri and sends the INVITE to C: INVITE C@domain2 SIP/2.0 To: B@domain1 From: A@domain1
- The INVITE arrives to the inbound proxy of C, who rewrites the Request-Uri with the location of C AoR and sends the INVITE to it: INVITE C@ip_c SIP/2.0 To: B@domain1 From: A@domain1
When the INVITE arrives to C it knows that the original destination was B. In conclusion, the "To" header is just valid for this purpose and no more. It has NOTHING to do with the real destination of the call (except in the original INVITE in which "To" and RURI are the same).
Hope it helps ;)
Regards.
Hi Inaki,
I am working on understand it thoroughly. Thank you very much for your help.
Best regards, Shie
Quoting Iñaki Baz Castillo ibc@aliax.net:
El Sábado, 15 de Marzo de 2008, shiebar@post.tau.ac.il escribió:
Hi Inaki,
I have already read it but probably missed this important detail. Thanks for the advice. I will read it again more carefully.
In resume: The real destination is the Request-Uri (except if there is a "Route" header). "To" header jist means the **original** destination. For example:
- A calls B so it generates an INVITE like: INVITE B@domain1 SIP/2.0 To: B@domain1 From: A@domain1
(note Request-Uri = "To" header)
A sends the INVITE to the proxy responsible for domain1.
The proxy has has a diversion to C@domain2 for B.
Proxy rewrites the request-Uri and sends the INVITE to C: INVITE C@domain2 SIP/2.0 To: B@domain1 From: A@domain1
The INVITE arrives to the inbound proxy of C, who rewrites the Request-Uri
with the location of C AoR and sends the INVITE to it: INVITE C@ip_c SIP/2.0 To: B@domain1 From: A@domain1
When the INVITE arrives to C it knows that the original destination was B. In conclusion, the "To" header is just valid for this purpose and no more. It has NOTHING to do with the real destination of the call (except in the original INVITE in which "To" and RURI are the same).
Hope it helps ;)
Regards.
-- Iñaki Baz Castillo
Users mailing list Users@lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users
El Sábado, 15 de Marzo de 2008, Iñaki Baz Castillo escribió:
El Sábado, 15 de Marzo de 2008, shiebar@post.tau.ac.il escribió:
Hi Inaki,
I have already read it but probably missed this important detail. Thanks for the advice. I will read it again more carefully.
Look at my previous mail (the pure mail headers):
From: =?utf-8?q?I=C3=B1aki_Baz_Castillo?= ibc@aliax.net To: users@lists.openser.org
But the fact is that the mail arrived to you !!!! XDD This is because during the SMTP protocol, the destination user for that mail was changed in favour of your user, but in the "To" headers remains the *original* destination (just for informative purposes).
It's same as in SIP.
;)
Hi Inaki,
Thanks for your last help. I have found that I didn't add the rule for INVITE in my sip.xml. Beginners mistakes :-). Now when trying to make a call I get no route exception: StandardProxy [SipProcessor[4]] - javax.sip.TransactionUnavailableException: no route!
Can you please help with this issue (or at least give my some direction were to look for the problem).
Thanks in advance, Shie Quoting Iñaki Baz Castillo ibc@aliax.net:
El Sábado, 15 de Marzo de 2008, Iñaki Baz Castillo escribió:
El Sábado, 15 de Marzo de 2008, shiebar@post.tau.ac.il escribió:
Hi Inaki,
I have already read it but probably missed this important detail. Thanks for the advice. I will read it again more carefully.
Look at my previous mail (the pure mail headers):
From: =?utf-8?q?I=C3=B1aki_Baz_Castillo?= ibc@aliax.net To: users@lists.openser.org
But the fact is that the mail arrived to you !!!! XDD This is because during the SMTP protocol, the destination user for that mail was changed in favour of your user, but in the "To" headers remains the *original* destination (just for informative purposes).
It's same as in SIP.
;)
-- Iñaki Baz Castillo
Users mailing list Users@lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users
El Thursday 27 March 2008 16:24:45 shiebar@post.tau.ac.il escribió:
Hi Inaki,
Thanks for your last help. I have found that I didn't add the rule for INVITE in my sip.xml. Beginners mistakes :-). Now when trying to make a call I get no route exception: StandardProxy [SipProcessor[4]] - javax.sip.TransactionUnavailableException: no route!
Can you please help with this issue (or at least give my some direction were to look for the problem).
Sorry, I've no idea about WeSIP or Java :(
-
Iñaki Baz Castillo -
Iñaki Baz Castillo -
shiebar＠post.tau.ac.il