Re: [OpenSIPStack] B2BUA how to route

[Joegen E. B. <joe...@gm...>]

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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