[SIPForum-techwg] SIPConnect 1.1 - Microsoft proposalandadditional requirements- TCP

Wed May 14 20:09:36 EDT 2008

Yeah, what I was alluding to is the following:

*	
	Basically, RFC 3261 says that you should switch to TCP "on the fly" whenever you send a request or a response that is too big. Practically speaking, this is extremelly difficult to implement and I have never heard of any implementation actually doing this. Furthermore, this concept of "upgrading on the fly to TCP from TCP" whenever a packet is too big plainly does not work in many scenarios. For example, it will not work when there is a NAT or firewall because bindings and pinholes are assigned on the 5-tuple not the 3-tuple (i.e., creating a binding or opening a pinhole with UDP does NOT create the corresponding binding or open the corresponding pinhole for TCP). 
*	
	What I am therefore saying is that if there is any chance that messages MIGHT be bigger than the maximum, then you MUST use TCP for all SIP signalling, as opposed to attempt this "on the fly upgrading" that basically doesn't work.

Another note on "simple devices" that don't send big messages... While this may be true, you never know what the far end will send you. The messages from the far end may be big. And in fact (this is a trunking specification), they often are.

If you go to a few scenarios in your head, you'll find out that UDP will break in many scenarios.

Finally, remember supporting TCP is mandatory in RFC 3261...

________________________________

	From: Peter Dunkley [mailto:peter at dunkley.me.uk] 
	Sent: Wednesday, May 14, 2008 16:03
	To: Audet, Francois (SC100:3055)
	Cc: Zweig, Greg; Joanne McMillen; Russell Bennett; techwg at sipforum.org
	Subject: Re: [SIPForum-techwg] SIPConnect 1.1 - Microsoft proposalandadditional requirements- TCP

	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 

					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

					----- Original Message ----- 

					From: Russell Bennett <mailto:Russell.Bennett at microsoft.com>  

					To: 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 is in a call with Bob at y.com, she is unlikely to want to transfer Bob to 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

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