[SIPForum-techwg] SIPConnect 1.1 - Microsoft proposalandadditional requirements- TCP
Peter Dunkley
peter at dunkley.me.uk
Wed May 14 19:03:28 EDT 2008
I am not sure there is a need to specify that TCP MUST be used when
packets will fragment as this is already covered in RFC 3261.
Section 18.1.1 Sending Requests clearly indicates that if a request is
within 200 bytes of the path MTU, or if the request is larger than 1300
bytes and the path MTU is unknown, the request MUST be sent over a
congestion controlled transport protocol (in this case TCP).
I do not favour removing the option of using UDP. For simple devices
that will not generate large messages there is some advantage in
allowing UDP to be used and no apparent benefit from using TCP. From a
purely engineering point of view, an operating system and TCP/IP stack
is still not guaranteed in a (simple) embedded device, and in the
absence of these it is almost trivial to implement (non-fragmenting)
UDP/IP, but certainly not so for TCP/IP. From a practical point of view
there are still a number of implementations that only support UDP
(though this is getting less common), which can be seen in the SIPit
implementation survey results.
I am not sure that RECOMMENDing TCP in all cases will have an effect.
If UDP is allowed there will be no shortage of implementations that use
it in preference if that suits their implementation and usage scenario -
regardless of what is recommended.
The RFC also indicates that while you should not fragment, you must
support receipt of fragmented messages. However, is this something that
SIPconnect could help with, by specifically making proof of compliance
to section 18.1.1 mandatory. This would ensure that a SIPconnect
compliant device would never use UDP inappropriately (but still cope
with devices that do).
Regards,
Peter
Francois Audet wrote:
> Right.
>
> That's why I was thinking that one option is to describe WHEN you MUST
> use TCP. Those would include:
>
> *
> When you need to secure the signaling chanel with TLS (obviously)
> *
> When it is possible that any packets will be so big they will
> fragment and cause problems
>
> And also, RECOMMEND to use TCP otherwise.
>
>
> ------------------------------------------------------------------------
> *From:* Zweig, Greg [mailto:gzweig at sonusnet.com]
> *Sent:* Wednesday, May 14, 2008 14:58
> *To:* Audet, Francois (SC100:3055); Joanne McMillen; Russell
> Bennett; techwg at sipforum.org
> *Subject:* RE: [SIPForum-techwg] SIPConnect 1.1 - Microsoft
> proposalandadditional requirements- TCP
>
> I would encourage the group to look at the issue beyond the
> technical merits. In my mind SIPConnect is all about encouraging
> interoperability; we all benefit from the rapid adoption of this
> paradigm. Based on my experience, I would suspect that there is a
> significant installed base of IP-PBXs that can't support a SIP
> over TCP solution without a significant software upgrade and
> potentially large hardware investments. I'm also concerned that
> some vendors will delay adoption when they are forced to upgrade
> portfolio elements to make their solutions compliant. Worse, they
> could burden solutions with convoluted translation devices.
> Making a change like this can have significant "ripple" effects
> that we should not discount. It would seem that Microsoft has
> made a wise strategic decision with their architecture but I think
> we have to recognize that this is a new design and we could be
> unintentionally burdening businesses that took our advice and
> moved to a VoIP solution so they would be "ready" for the future.
> I don't want to stop time but I think we'll more likely need a
> dual solution that accommodates either in some way. Perhaps It
> might mean two levels of compliance (like b and g in wireless)?
> I'm not going to suggest how but I think the goal needs to be
> considered.
>
>
>
> I would also like to understand Microsoft's plans for RT Codec
> support? I'm a big fan of wideband codecs such as G722.1 and RT
> so I believe such solutions should be included but it seems that
> more common narrowband codecs like G.729 need to be there. Most
> of the IP phones older than 14-18 months can't support G.722.1
> and most are already licensed for 729.
>
>
>
> Cheers,
>
>
>
>
>
> Greg Zweig
>
> Sonus Networks. Inc
>
> (978) 614-8027
>
>
>
> ------------------------------------------------------------------------
>
> *From:* techwg-bounces at sipforum.org
> [mailto:techwg-bounces at sipforum.org] *On Behalf Of *Francois Audet
> *Sent:* Wednesday, May 14, 2008 5:04 PM
> *To:* Joanne McMillen; Russell Bennett; techwg at sipforum.org
> *Subject:* Re: [SIPForum-techwg] SIPconnect 1.1 - Microsoft
> proposalandadditional requirements
>
>
>
> Right. We talked about defining draft-ietf-sip-outbound to work
> only with TCP, but people didn't want to do this.
>
>
>
> I don't expect a formal deprecation of UDP until there is a SIP
> 3.0, which there are no plans for today.
>
>
>
> ------------------------------------------------------------------------
>
> *From:* Joanne McMillen [mailto:joanne at avaya.com]
> *Sent:* Wednesday, May 14, 2008 14:00
> *To:* Audet, Francois (SC100:3055); Russell Bennett;
> techwg at sipforum.org
> *Subject:* Re: [SIPForum-techwg] SIPconnect 1.1 - Microsoft
> proposal andadditional requirements
>
> I agree with the proposal as well for mandating TCP in the
> Forum - just wanted to know if anyone had the answer to
>
> the IETF question since we will indeed be "a bit
> non-compliant" with 3261 when making the mandate...
>
>
>
>
>
> ----- Original Message -----
>
> *From:* Francois Audet <mailto:audet at nortel.com>
>
> *To:* Joanne McMillen <mailto:joanne at avaya.com> ; Russell
> Bennett <mailto:Russell.Bennett at microsoft.com> ;
> techwg at sipforum.org <mailto:techwg at sipforum.org>
>
> *Sent:* Wednesday, May 14, 2008 1:47 PM
>
> *Subject:* RE: [SIPForum-techwg] SIPconnect 1.1 -
> Microsoft proposal andadditional requirements
>
>
>
> I dont' think that it will be deprecated anytime soon.
>
>
>
> That being said, there are things like TLS that require
> TCP, and things like draft-ietf-sip-outbound (for high
> availability and NAT traversal) that work a lot better
> with TCP than with UDP.
>
>
>
> So I'm thinking that a "slow migration" is more likely.
>
>
>
> I don't think it would be out of the question for SIP
> Forum to mandate use of TCP if we wanted to. Or recommend
> it. Or explain that it must be use for certain scenarios,
> etc. We have many options.
>
>
>
> ------------------------------------------------------------------------
>
> *From:* techwg-bounces at sipforum.org
> [mailto:techwg-bounces at sipforum.org] *On Behalf Of
> *Joanne McMillen
> *Sent:* Wednesday, May 14, 2008 13:36
> *To:* Russell Bennett; techwg at sipforum.org
> *Subject:* Re: [SIPForum-techwg] SIPconnect 1.1 -
> Microsoft proposal andadditional requirements
>
> One quick question regarding number two:
>
>
>
> Wasn't it the case that IETF was going to deprecate
> UDP in 3261? What happened to that?
>
>
>
> This debate has been going on for years - perhaps the
> Forum could influence getting resolution there in IETF?
>
>
>
> --------------------------------------
> Joanne McMillen
> Systems Engineer
> Converged Communications Division
> Avaya Inc.
> Phone/Fax: +1.303.538.4060
> Email: joanne at avaya.com <mailto:joanne at avaya.com>
>
>
>
>
>
> ----- Original Message -----
>
> *From:* Russell Bennett
> <mailto:Russell.Bennett at microsoft.com>
>
> *To:* techwg at sipforum.org
> <mailto:techwg at sipforum.org>
>
> *Sent:* Wednesday, May 14, 2008 12:58 PM
>
> *Subject:* [SIPForum-techwg] SIPconnect 1.1 -
> Microsoft proposal andadditional requirements
>
>
>
> All,
>
>
>
> The list has been very quiet since we submitted
> our proposal, so let me start the discussion.
>
>
>
> First of all, here some clarification /
> explanation of the origin of the document.
>
>
>
> This **is** (as many of you will probably suspect)
> a **largely** neutralized version of a spec that
> we developed for our own purposes. We have been
> working with Service Providers to come up with an
> interface definition that they and we can support
> for SIP Trunking. We have reviewed that document
> with a number of Service Providers and most (but
> not all) have found no issue with our approach.
>
>
>
> I was asked by Rich Shockey and Scott Hoffpauir if
> we would be willing to submit that document to the
> TWG to "prime the pump" for SC1.1. Our initial
> goal is to ease deployment of SIP Trunk solutions
> for everyone -- so we were happy to do this. The
> sections of the document that are Microsoft
> specific will be altered as a natural outcome of
> this process.
>
>
>
> I have received private feedback internally and
> externally that the document is incomplete: of
> this I have no doubt. We, like everyone else, are
> limited to a view of the world from our own
> perspective -- this is where collaboration among a
> group of well informed peers will rapidly drive a
> clean and easy to deploy standard.
>
>
>
> I strongly suggest that we don't try to boil the
> ocean with myriad issues and requirements, albeit
> valid, that will go much beyond basic "call setup
> and teardown". Our objective should be to
> develop a standard that makes it easy for vendors
> and service providers to adopt SC1.1. We can
> "layer on" additional requirements in later
> versions of the standard that can be selectively
> adopted as required.
>
>
>
> That being said, let me comment on particular
> requirements that have been brought to my attention:
>
>
>
> 1) Definition of network elements:
>
>
>
> I have anonimized the elements that interface
> between two networks as "Enterprise Proxy" and
> "Service Provider Proxy". These are logical
> elements that are capable of handling signaling
> and media. There is no attempt to suggest or
> limit any capabilities of these elements beyond
> establishing voice sessions between two networks.
> The bundling of additional value in these
> elements, including security features, is IMHO
> outside the scope of this effort. Enterprises,
> Vendors and Service Providers are at liberty to
> define their own features and requirements that go
> beyond the transmission of voice traffic.
>
>
>
> 2) SIP Transport:
>
>
>
> I proposed UDP=MAY, TCP=MUST, TLS=MAY. While this
> does not align perfectly with 3261, there is a
> strong argument for TCP as the base transport
> based on the fact that SIP messages tend to exceed
> 1500 bytes these days. 3261 says that a UA
> offering UDP must also offer TCP, but not vice versa:
>
>
>
> Making TCP mandatory for the UA is a substantial
> change from RFC 2543. It has arisen out of the
> need to handle larger messages, which MUST use
> TCP, as discussed below. Thus, even if an element
> never sends large messages, it may receive one and
> needs to be able to handle them.
>
>
>
> Many current SPs deviate from this, offering only
> UDP on the grounds that TCP is too resource
> intensive. I believe that this can be resolved
> with TCP connection reuse. Furthermore, "service
> provider proxies" in deployment today are capable
> of B2B'ing TCP to UDP -- so this issue can be
> overcome via a choice that they are well able to
> implement.
>
>
>
> IMHO, it is too early to make TLS (or SRTP for
> that matter) a MUST, although this is a desirable
> longer term goal. There are many issues to be
> worked out with respect to this, and we can layer
> communications security requirements onto the
> standard later. For now, most (all?) SPs offer
> SIP Trunking on a closed data connection with a
> VPN, so resolving that issue is not urgent.
>
>
>
> 3) 911/emergency calling, including provision
> of location information
>
>
>
> This is a regulatory requirement in some
> jurisdictions but not others, therefore it should
> be optional and arguably, covered in a later
> spec. The advent of mobile IP communications has
> made this a difficult requirement to resolve
> 'automagically'. There are vendors who attempt
> to resolve it but there is no clear standard. At
> Microsoft, we are working on this, but have
> nothing to contribute to a standard at this time.
>
>
>
> 4) FAX
>
>
>
> This is an area that c/should be added to the
> discussion.
>
>
>
> 5) List of codecs to be offered
>
>
>
> Based on discussions with SPs, we understand that
> G.711 is not universally acceptable due to
> bandwidth utilization concerns and that G.722.1
> (for example) would be a better choice. I
> suggest that we limit the mandatory set of codecs
> to the minimum set of universally acceptable
> codecs and not mandate codecs that have onerous
> IP/royalty conditions attached to them.
>
>
>
> 6) Privacy
>
>
>
> This is a hot topic and various RFCs address
> this. We take a very strict view of privacy that
> is implemented within our infrastructure,
> regardless of what is sent in the SIP message.
> However, what is under consideration in this
> initiative is the definition of private SIP/RTP
> connection between a SP and an Enterprise (i.e.
> the sessions do not traverse untrusted
> intermediary network elements or public
> networks). Therefore, I suggest that we defer
> privacy issues to a later version of the spec and
> assume for now that privacy is addressed in the
> trust relationship between those two entities.
>
>
>
> 7) Other features, e.g. Transfer, Call
> History, etc.
>
>
>
> These are 'nice-to-haves' that can be dealt with
> later in order to speed completion of this
> initiative. I suspect that support of these
> mechanisms is spotty and this will be an adoption
> blocker in the short term. Per above, we should
> address the uber issue of SC1.0 non-adoption by
> making SC1.1 easy to adopt. For example: it is
> not clear to me that Transfer is mandatory
> requirement in a SIP Trunk: this function should
> be handled by elements behind the Enterprise and
> Service Provider Proxies. If Alice at x.com
> <mailto:Alice at x.com> is in a call with Bob at y.com
> <mailto:Bob at y.com>, she is unlikely to want to
> transfer Bob to Carol at y.com <mailto:Carol at y.com>.
>
>
>
> Russell
>
>
>
> ---------------------------------------------------
>
> Russell Bennett
>
> Partner Program Manager
>
> Office Communications Group
>
>
>
> Microsoft Corporation
>
> One Microsoft Way
>
> Redmond, WA 98052-6399
>
>
>
> mailto: russben at microsoft.com
> <mailto:russben at microsoft.com>
>
> sip: russben at microsoft.com
> <mailto:russben at microsoft.com>
>
> tel: +1 (425) 706-3622
>
> http://www.microsoft.com/uc/
>
>
>
> ------------------------------------------------------------------------
>
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--
*Peter Dunkley*
*Email:* peter at dunkley.me.uk <mailto:peter at dunkley.me.uk>
*http://www.linkedin.com/in/pdunkley*
*My Website <http://www.dunkley.me.uk/>*
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