Re: [OpenSIPStack] B2BUA how to route
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Re: [OpenSIPStack] B2BUA how to route
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From: Joegen E. B. <joe...@gm...> - 2007-12-13 02:46:27
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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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>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
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>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>>
>>>
>>>
>>>
>>>
>>
>
>
>
>
>
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