Hi everybody ! I am trying to create a basic B2BUA server using nist-sip stack, "based" on existing presence-proxy. This server will be used as an example of SIP call establishment authorization: 2 SIP softphones configured to use my server for registar/proxy. I am trying to modify nist presence-proxy to have a B2BUA behavior.
My problem is that I can't find an document (other than rfc3261) describing the behavior of such a server (and such a use), especially to have a better undestanding on (what I call) request and response forwarding. Of course my two phones register without problem because the dialog is only between server and UAC. First question: is it right to configure a SIP phone to use my server registrar/proxy: it is a registrar/b2bua indeed. Is there any problem to act like this ? Second question: My first phone (phone1) sends an INVITE like this one: .INVITE sip:[EMAIL PROTECTED]:6500 SIP/2.0. .Call-ID: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> .CSeq: 1 INVITE. .From: "GuiPoM" <sip:[EMAIL PROTECTED]>;tag=2294486625. .To: <sip:[EMAIL PROTECTED]>. .Via: SIP/2.0/UDP 127.0.0.1:4000;branch=z9hG4bK39102966f9e82947ebc9637a4d9e3f8e. .Max-Forwards: 20. .Contact: <sip:127.0.0.1:4000;transport=udp>. .Content-Length: 0.... It is received by B2BUA server because of VIA or Contact field. B2BUA must create another request based on this one to contact second SIP phone (phone2). Which fields have to be modified ? I guess that From has to be changed to avoid that phone2 directly contact phone1 without using B2BUA. But how phone2 will be able to know who is calling if this field is modified ? Via must be the same, but maybe whith another branch. CallID must be changed, because this is another dialog. Assuming everything is ok, phone2 gives an aswer to B2BUA. This implies that on my server, an association between to UAC has been created to "forward" the request. Sorry if my questions are quite common, but I can't find examples of B2BUA call flows (with headers' values) and I am quite a beginner as a SIP implementor ;) Don't forget that I am just trying to realize something very simple with only REGISTER, INVITE, OK, ACK and trying/ringing methods and not fully compliant with all SIP's requirements. I also had a look to the nist-sip 3pcc project, which should be quite close to a b2bua, but INVITEs are generated by server and this is helpless for my actual problems. Thanks for your help Guillaume _______________________________________________ Sip-implementors mailing list [email protected] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
