On Sat, Feb 7, 2009 at 1:05 PM, Paolo Prandini <[email protected]> wrote:
> Hi all! I would like to help with this development, could you
> please suggest me where I could start looking in the sources
> for this fix?
> Thanks

Its somewhat detailed and not included in our current release.
Unfortunately, I have to stay focused on getting the current release
completed before we look at this.

Regards

Ranga

>
> M. Ranganathan ha scritto:
>>
>> On Mon, Jan 26, 2009 at 4:27 PM, Paolo Prandini <[email protected]> wrote:
>>>
>>> Before considering sipx we checked other solutions, but SBCs
>>> are not meant to be used as B2BUA and sip proxies. In fact
>>> we don't need nor want media relay and that's the main
>>> purpose of SBCs, isn't it? I think sipx could be quite the
>>> solution we're looking for, apart from writing our own.
>>
>>
>> The reason for a relay is because of the requirement to handle
>> hairpinned call transfers from behind a NAT ( i.e. transfer agent
>> behind NAT but both transferee and transfer target on the public
>> network ).
>>
>> We probably cannot fit it into this release ( I think ) but possibly
>> what you need can be prototyped and tested in your environment  --
>> especially because call transfer is not a requirement for you.
>>
>> I have created a JIRA issue for this :
>> http://track.sipfoundry.org/browse/XECS-2140
>>
>> Please watch it for updates and please comment if needed.
>>
>> Incidentally, while media relaying should be removed from the media
>> path when not needed, I believe (based on experiments) that sipxbridge
>> can handle well more than a "couple of dozen connections".
>>
>> Ranga
>>
>>
>>>
>>>> Overall, sipXbridge was not meant as a full scale SBC. In your instance,
>>>> you might look at OSBC or something like that which will give you (I
>>>> think) what you are looking for. I don't recall sipxbridge being meant
>>>> to
>>>> be used in a solution with more than a couple of dozen connections, at
>>>> least in its current form.
>>>>
>>>>>>> "Paolo Prandini" <[email protected]> 01/26/09 4:09 PM >>>
>>>>
>>>> I agree. We won't use call transfer anyway.
>>>>
>>>>> Paolo Prandini wrote:
>>>>>>
>>>>>> Yes I understand.
>>>>>> My use case is quite special, I know.
>>>>>> We are a small ISP and deliver also VoIP services, that we
>>>>>> buy from bigger operators. The problem is that our services
>>>>>> are quite reliable, but bigger operators'ones are not...so
>>>>>> customers complain with us of service interruptions.
>>>>>> We would like to have CPE ( no NAT involved anyway ) register
>>>>>> with us ( to sipx), so we can direct outgoing calls to whatever
>>>>>> operator is working/cheapest at the moment, while receiving
>>>>>> calls from the right DIDs. But we don't want to be in the middle
>>>>>> of the media path otherwise we need a big amount of quite good
>>>>>> bandwidth...
>>>>>> Thanks anyway
>>>>>> Paolo
>>>>>>
>>>>>>
>>>>> Correct me if I am wrong but it appears that for this application you
>>>>> would not need to support call transfer ( i.e. REFER translation to
>>>>> re-INVITE).
>>>>>
>>>>>
>>>>> If so, this makes the problem much simpler.
>>>>>
>>>>> Ranga
>>>>>
>>>>>> Scott Lawrence ha scritto:
>>>>>>
>>>>>>> On Mon, 2009-01-26 at 15:00 +0100, Paolo Prandini wrote:
>>>>>>>
>>>>>>>> I managed to use the sipXbridge, very nice indeed.
>>>>>>>> I am however in the following case:
>>>>>>>> a) sipx server has a public ip, no NAT
>>>>>>>> b) sip trunk has a public ip, no NAT
>>>>>>>> c) users are on the internet with static public IP, no NAT
>>>>>>>> In this case I expect RTP to be between sip trunk and
>>>>>>>> users without media server intervention, packets should
>>>>>>>> be flowing directly between endpoints without sipx in
>>>>>>>> the middle.
>>>>>>>> What I find is that sipx stays in the media path, and
>>>>>>>> that is clearly undesiderable.
>>>>>>>> How can I obtain this result?
>>>>>>>>
>>>>>>> I might not be quite as undesirable as you think.
>>>>>>>
>>>>>>> First of all, the sipXrelay that's actually passing the RTP packets
>>>>>>> is
>>>>>>> very quick and doesn't do much harm (or any transcoding).  I owe the
>>>>>>> list a posting on this, actually...
>>>>>>>
>>>>>>> The sipXbridge exists to manage your interface to the provider,
>>>>>>> especially authentication.  Some providers will require that the SDP
>>>>>>> be
>>>>>>> coming from the same place the SIP came from (yes, this is a silly
>>>>>>> requirement, but...), and there are other problems that sipXbridge
>>>>>>> can
>>>>>>> take care of for you - like the fact that most providers can't deal
>>>>>>> with
>>>>>>> REFER.
>>>>>>>
>>>>>>> That having been said, it may be that sipXbridge could detect that no
>>>>>>> NAT compensation is needed and get the sipXrelay out of the way if it
>>>>>>> were configured to do so.  Ranga - can this be done now?  If not, we
>>>>>>> could put in an improvement issue for some future release...
>>>>>>>
>>>>>>>
>>>>>>>
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>>>>>
>>>>
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>>>>
>>>
>>>
>>
>>
>>
>
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> Dr.Ing.Paolo Prandini
>
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-- 
M. Ranganathan
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