Re: [OpenSIPStack] B2BUA how to route

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Hi Joegen

Where is the SIPHeader header located?

I'm putting the

> SIPHeader myHeader(  "Remote-Party-Id", "Whatever the value is" );
> invite.AddCustomHeader( myHeader );

code in  SBCBackDoorTrunk.cxx. inside of the
SBCBackDoorHandler::SBCBackDoorHandler method..  This should call the
constructor first and execute the
> SIPHeader myHeader(  "Remote-Party-Id", "Whatever the value is" );
> invite.AddCustomHeader( myHeader );
Next.

Unless you think it should go somewhere else

Warren Kreckler

----- Original Message ----- 
From: "Joegen E. Baclor" <joe...@gm...>
To: "sales@ER" <sa...@el...>;
<ope...@li...>; "listaopenSBC"
<ope...@li...>
Sent: Wednesday, December 12, 2007 8:46 PM
Subject: Re: [OpenSIPStack] B2BUA how to route

> sales@ER wrote:
> > Hi Joegen
> >
> >
> >> I see what you mean.   I am not really familiar with the use of the
> >> Remote-Party-Id.   We have implemented P-Asserted-Identity for this
> >> instead.   Can you point me to the RFC that discusses the use cases for
> >> Remote-Party-Id?
> >>
> >
> > Yes the P-Asserted-Identity replaced the Remote-Party-Id in the RFC but
it
> > is still in used in older SBC models and my ITSP has not updated the SBC
i
> > am accessing.  I need to modifry the xml code to and and elseif to test
for
> > this possiblity.   Please direct me to the xml that is managing this
> > identity.
> >
>
> For you be able to rewrite any header before it gets sent to the UAS you
> need to override SBCBackDoorCallHandler::OnOutgoingCall();
>
> Look for the declaration of class SBCBackDoorCallHandler in
> SBCBackDoorTrunk.cxx.  Add a new member function
>
> virtual void OnOutgoingCall(
>     B2BUAConnection & connection,
>     B2BUACall & call,
>     SIPMessage & invite
>   );
>
>
> This function will be called whenever there is a new INVITE that will be
> sent out by the backdoor trunk.   Implement this function right after
> BOOL SBCBackDoorCallHandler::OnReceivedMergedInvite() methid in
> SBCBackDoorTrunk.cxx
>
>
> You may add special headers to invite using this code
>
> SIPHeader myHeader(  "Remote-Party-Id", "Whatever the value is" );
> invite.AddCustomHeader( myHeader );
>
> HTH
>
> Joegen
> > Warren Kreckler
> >
> >
> > ----- Original Message ----- 
> > From: "Joegen E. Baclor" <joe...@gm...>
> > To: "sales@ER" <sa...@el...>
> > Sent: Tuesday, December 11, 2007 8:32 PM
> > Subject: Re: [OpenSIPStack] B2BUA how to route
> >
> >
> >
> >> sales@ER wrote:
> >>
> >>> Hi Joegen
> >>>
> >>> Thank you very much for your replies.
> >>>
> >>> 1.  I'm using the lastest version.
> >>>
> >>>
> >> Then your ITSP must be seeing just a single via.  If you think the
> >> contrary, send me a packet capture from sipx->OpenSBC and OpenSBC->ITSP
> >>
> >>
> >>
> >>
> >>> 3.  sipX does not re-write  header as far as I know.  Are you asking
for
> >>> sipX header(s) dealing with Caller-ID?
> >>>
> >>>  Remote-Party-ID to determine the Calling ID.  This is not an element
> >>> created
> >>>  by Sipx.  The SBC will need to extract the user part of the From URI
> >>>
> > and
> >
> >>>  create a Remote-Party-ID. I did not see this capability with OpenSBC.
> >>>  Without this, the called party on the PSTN will either see "Private
> >>>  Caller"or "Anonymous" on their phone instead of the DID.
> >>>
> >>>
> >>>
> >> I see what you mean.   I am not really familiar with the use of the
> >> Remote-Party-Id.   We have implemented P-Asserted-Identity for this
> >> instead.   Can you point me to the RFC that discusses the use cases for
> >> Remote-Party-Id?
> >>
> >>
> >>
> >>> Warren Kreckler
> >>>
> >>>
> >>>
> >>>
> >>>
> >>> ----- Original Message ----- 
> >>> From: "Joegen E. Baclor" <joe...@gm...>
> >>> To: "sales@ER" <sa...@el...>
> >>> Cc: <ope...@li...>
> >>> Sent: Sunday, December 09, 2007 7:21 PM
> >>> Subject: Re: [OpenSIPStack] B2BUA how to route
> >>>
> >>>
> >>>
> >>>
> >>>> inline...
> >>>>
> >>>> sales@ER wrote:
> >>>>
> >>>>
> >>>>> Yes  They call it peer to peer.  By that they meam
> >>>>>
> >>>>>
> >>>>> 1. Via Headers: ITSP has stated that they can accept only 1 Via
> >>>>> statement in an INVITE.  As background, each device will add a Via
> >>>>>
> >>>>>
> >>> statement
> >>>
> >>>
> >>>>> to the INVITE to if it has processed the INVITE.  Only the last or
top
> >>>>>
> >>>>>
> >>> entry
> >>>
> >>>
> >>>>> is really of interest to the party that next handles the INVITE.  In
> >>>>>
> >>>>>
> >>> order
> >>>
> >>>
> >>>>> for ITSP to accept the INVITE of an outbound call, OpenSBC will
> >>>>> need to strip off all previous Via statements from the INVITE and
add
> >>>>>
> >>>>>
> >>> its'
> >>>
> >>>
> >>>>> own.  I have not found any capability to remove the previously
> >>>>>
> > inserted
> >
> >>> Via
> >>>
> >>>
> >>>>> statements.
> >>>>>
> >>>>>
> >>>>>
> >>>> What version are you using?   There was a bug introduced when we got
> >>>> back from sipIT 21 due to the changes made there that had the vias
not
> >>>> getting stripped.  Please use the latest CVS.   OpenSBC should be
> >>>> stripping the via before the B2BUA sends the INVITE out to the UAS.
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>> 2.  Lock IP Address and port to first sender: This option comes into
> >>>>>
> >>>>>
> >>> play
> >>>
> >>>
> >>>>> when a call has been answered either by a person or system component
> >>>>>
> >>>>>
> >>> (i.e.
> >>>
> >>>
> >>>>> Auto Attendant) and a transfer is attempted.  When the transferred
> >>>>>
> > call
> >
> >>> is
> >>>
> >>>
> >>>>> answered by a new phone or component, it will negotiate use of a new
> >>>>>
> > RTP
> >
> >>>>> port for the media stream.  Some service providers, ITSP included,
> >>>>> do not allow the RTP port to change once the initial call is
> >>>>>
> >>>>>
> >>> established.
> >>>
> >>>
> >>>>> They do this to protect against the "hijacking" of a call by
Hackers.
> >>>>>
> >>>>>
> >>> Since
> >>>
> >>>
> >>>>> the media is flowing through a SBC, the SBC then needs to manage
which
> >>>>>
> >>>>>
> >>> ports
> >>>
> >>>
> >>>>> are used to exchange media (voice).   If the original port is not
> >>>>>
> >>>>>
> >>> utilized
> >>>
> >>>
> >>>>> for the media back to the carrier, the PSTN will not hear any audio
> >>>>>
> > once
> >
> >>> the
> >>>
> >>>
> >>>>> call is transferred.  I do not see this capability with OpenSBC.
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>> In media proxy mode (Always Proxy Media = true), OpenSBC does not
> >>>>
> > change
> >
> >>>> the port of RTP even during reInvites.
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>> 3.  Calling ID:  SIPxchange utilizes the From: element to provide
the
> >>>>> Calling ID (DID).  It normally inserts the userID in the user part
of
> >>>>>
> >>>>>
> >>> the
> >>>
> >>>
> >>>>> >From URI.  ITSP uses the INVITE element
> >>>>> Remote-Party-ID to determine the Calling ID.  This is not an element
> >>>>>
> >>>>>
> >>> created
> >>>
> >>>
> >>>>> by Sipx.  The SBC will need to extract the user part of the From URI
> >>>>>
> > and
> >
> >>>>> create a Remote-Party-ID. I did not see this capability with
OpenSBC.
> >>>>> Without this, the called party on the PSTN will either see "Private
> >>>>> Caller"or "Anonymous" on their phone instead of the DID.
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>> Can you send a sample of this from header that is rewritten by sipX?
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>> Warren Kreckler
> >>>>>
> >>>>> ----- Original Message ----- 
> >>>>> From: "Joegen E. Baclor" <joe...@gm...>
> >>>>> To: <ope...@li...>
> >>>>> Cc: <jo...@op...>
> >>>>> Sent: Friday, December 07, 2007 12:08 AM
> >>>>> Subject: Re: [OpenSIPStack] B2BUA how to route
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>>> You need to use the SIP Trunking capability of OpenSBC for this.
Do
> >>>>>> you need to authenticate calls with your ITSP?
> >>>>>>
> >>>>>>
> >>>>>> sales@ER wrote:
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>> Hi
> >>>>>>>
> >>>>>>> Almost have this puppy working.
> >>>>>>>
> >>>>>>> Sipx and opensbc generally well understood.
> >>>>>>>
> >>>>>>> Problem:
> >>>>>>>
> >>>>>>> When OSBC receives INVITE from sipX =>  ITSP,
> >>>>>>> OSBC route the INVITE back to sipX.
> >>>>>>>
> >>>>>>> We have two rules in the B2Bua route
> >>>>>>> [sip:202.100.2.23:5060] sip: 202.100.2.23  this goes to our ITSP
> >>>>>>> [sip:sipx.sip.net:5060] sip:sipx.sip.net            this point to
> >>>>>>>
> > our
> >
> >>>>>>>
> >>>>> sipx
> >>>>>
> >>>>>
> >>>>>
> >>>>>>> the missing rule/route?
> >>>>>>>
> >>>>>>> Where do you put the rule and what should the rule say to route
> >>>>>>>
> > INVITE
> >
> >>>>>>>
> >>>>> out
> >>>>>
> >>>>>
> >>>>>
> >>>>>>> to our ITSP?
> >>>>>>>
> >>>>>>> Warren Kreckler
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>
>
>>>> -----------------------------------------------------------------------
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