Re: [OpenSIPStack] B2BUA how to route

[<sa...@ER...>]

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.

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.

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.

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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