[SIPForum-techwg] Close off new comments to 1.0 effort on 12Noon EST Tommrrow
Eric Burger
eburger at sipforum.org
Thu Sep 25 10:19:19 EDT 2008
I like
> The SIPConnect 1.1 Technical Recommendation outlines an interface
> specification that enables direct connectivity between a SIP-
> enabled Service Provider network and a SIP-enabled Enterprise Network.
Do note, that as mentioned in bullet 3, if we cannot quickly come to
consensus on a topic that is PSTN versus SIP cloud termination, we
will err on PSTN termination for this version of SIPconnect. That is
why I'm not a fan of "or SIP Enterprise Applications", as that opens
the door for long, drawn-out ratholes on theories over practice.
Just a personal opinion.
AND, THANKS FOR SUBMITTING TEXT. MAY YOU BE AN EXAMPLE TO EVERYONE ON
THE LIST!!!
On Sep 25, 2008, at 8:04 AM, Dan York wrote:
> Eric,
>
> As much as I am a strong advocate of end-to-end connectivity and
> want to do everything possible to build the massive SIP-SIP
> interconnect, I do have to (gasp) agree with the esteemed Mr. Dolly
> here that the scope of SIPconnect *today* really needs to focus on
> Enterprise <-> SP connectivity. I agree with you that if we can get
> in features for SIP-SIP connectivity that don't bog us down, we
> should do so. However I do think we need to be crystal-clear that
> SIPconnect 1.1 is all about doing the "SIP trunking" from an
> enterprise to a SIP Service Provider.
>
> My point was that in the abstract, the Sept 22 document states:
> -------------------
> The SIPConnect 1.1 Technical Recommendation outlines an interface
> specification that enables direct connectivity between a SIP-
> enabled Service Provider network and a SIP-enabled Enterprise
> Network for the purpose of originating and/or terminating calls from
> the Public Switched Telephone Network (PSTN).
> -------------------
> while I would suggest it should read more like:
> -------------------
> The SIPConnect 1.1 Technical Recommendation outlines an interface
> specification that enables direct connectivity between a SIP-
> enabled Service Provider network and a SIP-enabled Enterprise Network.
> -------------------
> For perhaps MOST users of the SIPconnect 1.1 specification, the
> usage *will* be for PSTN connectivity. However, there are certainly
> circumstances where the usage may be for interconnection of a
> premise SIP system to a SIP cloud for other services such as
> providing applications. To give a specific example, I could see the
> case where a SIPconnect compliant IP-PBX wanted to be able to do a
> "SIP trunk" to Voxeo's (hopefully at that point SIPconnect-
> compliant) hosted SIP application platform. The call might flow to
> our application server, have some application executed and then flow
> back to the on-premise IP-PBX. All via SIP... and all without
> touching the PSTN.
>
> Maybe I'm getting too pedantic or nuanced here - and I don't mean to
> take us down a rathole - but I would suggest that the abstract of
> the document should not necessarily focus *exclusively* on PSTN
> connectivity.
>
> Perhaps another way to say it is:
> -------------------
> The SIPConnect 1.1 Technical Recommendation outlines an interface
> specification that enables direct connectivity between a SIP-
> enabled Service Provider network and a SIP-enabled Enterprise
> Network. Such connectivity may be for the purpose of originating and/
> or terminating calls from the Public Switched Telephone Network
> (PSTN) or for accessing SIP-based applications provided by the
> Service Provider.
> -------------------
> Again, I don't really want to create a rathole on the first
> paragraph - I just don't want to limit the SIPconnect 1.1 spec to
> PSTN connectivity.
>
> Hoping-we-can-rapidly-move-beyond-paragraph-one,
> Dan
>
>
> On Sep 25, 2008, at 7:45 AM, DOLLY, MARTIN C, ATTLABS wrote:
>
>> Eric,
>>
>> I strongly disagree. The enterprise customer that purchases these
>> equipment will interconnect to a service provider for services and
>> PSTN connectivity. The end-to-end principle did not quite work for
>> the IETF, so I do not se it working here. You need to take the
>> carrier into account, or this specification is not worth the paper
>> it is written on.
>>
>> Cheers,
>>
>> Martin
>>
>> From: techwg-bounces at sipforum.org [mailto:techwg-
>> bounces at sipforum.org] On Behalf Of Eric Burger
>> Sent: Thursday, September 25, 2008 12:40 AM
>> To: Dan York
>> Cc: techwg at sipforum.org
>> Subject: Re: [SIPForum-techwg] Close off new comments to 1.0 effort
>> on 12Noon EST Tommrrow
>>
>> Let us not forget: one of the founding tenents of the SIP Forum is
>> the end-to-end principle.
>>
>> Some guidelines I'd like to see, unless someone squawks loudly:
>> So long as doing something that works SIP-SIP does not bog us down
>> (i.e., we get it for free), I am all for it.
>> Moreover, we should not do anything in SIPconnect 1.1 that will be
>> an obvious impediment to SIPconnect 2.0, which will have all of the
>> cool SIP-SIP, presence, full video conferencing, and sens-o-rama
>> content negotiation.
>> However, if it looks like we are going to rathole on something that
>> just is not germane to getting SIPconnect 1.1 done, then I, for
>> one, would not object to the editor or chair to cut discussion and
>> defer that particular item to SIPconnect 2.0.
>>
>> This does not mean we cannot have a discussion. Likewise, as far as
>> I know, this is entirely a theoretical discussion, as I am not
>> aware of anything we are doing in SIPconnect 1.1 that prevents SIP-
>> SIP from just working.
>>
>>
>>
>> On Sep 23, 2008, at 12:00 PM, Dan York wrote:
>>
>>> Richard,
>>>
>>> I've not sent any specific comments before because many of the
>>> comments I did have were picked up by various parties and it seemed
>>> pointless to simply say "I agree".
>>>
>>> I will state, though, that I second the comments of John Elwell and
>>> others that my view of SIPconnect is that it should be a reference
>>> architecture for connecting a on-premise SIP system to a SIP Service
>>> Provider. Period. This connection may or may not be for PSTN
>>> origination or termination. That is certainly one of the reasons -
>>> and may be the primary reason for many people - but it's not the
>>> only
>>> reason. I think we're getting (slowly) to the point where there
>>> *are*
>>> valid cases of wanting to connect SIP systems to SIP Service
>>> Providers
>>> purely to link in to the existing SIP clouds.
>>>
>>> My other comment is that in section 7.1.6, I think we need to think
>>> through whether other SIP messages beyond INVITE and REGISTER do
>>> need
>>> authentication, etc. (For instance, I know of some usage of REFER
>>> which would seem to need that authentication.)
>>>
>>> I look forward to the discussion on Thursday. Overall, I think the
>>> document is looking great.
>>>
>>> Regards,
>>> Dan
>>>
>>> On Sep 22, 2008, at 2:59 PM, Richard Shockey wrote:
>>>
>>>> Tuesday September 23, 2008 in advance of our Face to Face meeting
>>>> on
>>>> Thursday and Friday in Denver.
>>>>
>>>> This does not mean we are closing off comments in general, or
>>>> afterwards,
>>>> but we want to limit the discussion points to what we have in
>>>> hand on
>>>> Tuesday.
>>>>
>>>> After we meet and reach some consensus on the direction of the
>>>> document
>>>> Spencer will issue a new draft and we can hopefully start the
>>>> process to
>>>> closure.
>>>>
>>>> Richard Shockey
>>>> Director, Member of the Technical Staff
>>>> NeuStar
>>>> 46000 Center Oak Plaza - Sterling, VA 20166
>>>> PSTN Office +1 571.434.5651
>>>> PSTN Mobile: +1 703.593.2683
>>>> <mailto:richard(at)shockey.us>
>>>> <mailto:richard.shockey(at)neustar.biz>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
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>>>> Send mail to: techwg at sipforum.org
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>>>
>>> --
>>> Dan York, CISSP, Director of Emerging Communication Technology
>>> Office of the CTO Voxeo Corporation dyork at voxeo.com
>>> Phone: +1-407-455-5859 Skype: danyork http://www.voxeo.com
>>> Blogs: http://blogs.voxeo.com http://www.disruptivetelephony.com
>>>
>>> Build voice applications based on open standards.
>>> Find out how at http://www.voxeo.com/free
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> techwg mailing list
>>> Send mail to: techwg at sipforum.org
>>> Unsubscribe or edit options at: http://sipforum.org/mailman/listinfo/techwg
>>
>
> --
> Dan York, CISSP, Director of Emerging Communication Technology
> Office of the CTO Voxeo Corporation dyork at voxeo.com
> Phone: +1-407-455-5859 Skype: danyork http://www.voxeo.com
> Blogs: http://blogs.voxeo.com http://www.disruptivetelephony.com
>
> Build voice applications based on open standards.
> Find out how at http://www.voxeo.com/free
>
>
>
>
>
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