[SIPForum-techwg] IP PBX / SP Interop Draft Version 5 proposededits (update)

Joanne McMillen joanne at avaya.com
Wed Mar 8 11:43:50 EST 2006


IP PBX / SP Interop Draft Version 5 proposed edits (update)Chris,

I agree with all your recommendations and also am OK with adding Rohan's points
(with the latest agreed to wording).

I could not access the preview draft 5 for some reason, but have an additional comment
that I'll make against draft 4 instead (since the text in 5 is probably the same).

Sections 6.1 and 6.2 - registration:
Granted, registration is not an enterprise requirement, but I'd like to see some wording
tightened up in section 6.2 such that it does not get turned into one in practice.
Can we make the following change:

    "SIP Application Servers MUST be prepared to accept (but MUST not require) registrations for any valid AOR that 
    the Service Provider has
    assigned to an Enterprise. This interface specification does not define any specific action that is triggered by a successful
    registration; however one possible use of this information might be to update the SIP Application Server's "default" IP
    address for the PBX."

Does anyone have a problem with that change or something similar?

Joanne
  ----- Original Message ----- 
  From: Chris Sibley 
  To: SIP Forum Tech WG 
  Sent: Friday, March 03, 2006 1:24 PM
  Subject: [SIPForum-techwg] IP PBX / SP Interop Draft Version 5 proposededits (update)


  Added voting item for the updates to the NAT / Firewall traversal section as requested (in BLUE).

  Thanks,

  --Chris

  _____________________________________________
  From: techwg-bounces at sipforum.org [mailto:techwg-bounces at sipforum.org] On Behalf Of Chris Sibley
  Sent: Friday, March 03, 2006 12:52 PM
  To: 'SIP Forum Tech WG'
  Subject: [SIPForum-techwg] IP PBX / SP Interop Draft Version 5 proposed edits

  IP PBX / SP Interop WG team members,

  Following are the "top" issues derived from the "master list" of 2/27 - 3/2 comments. Please give them a quick read and respond to each with a "yea or nay" vote on the recommended disposition.

  After this round of edits/discussion has been completed, I will publish a final list of issues still requiring discussion (also compiled from the "master list").

  Thanks!

  --Chris

  -----------------------

  RFC 3265 (Session Initiation Protocol (SIP)-Specific Event Notification) Support

  Ernst Horvath wrote:

  "I would like to see the following RFCs removed from the table in section 5: 3265, 3515, 3842, 3891, and 3892. The reason is that there are no use cases specified for these extensions in the rest of the document, and it is unreasonable to mandate global conformance to a generic mechanism like REFER or the event framework. (Which role is mandatory - subscriber, notifier or both - for which entity? Which methods can be referred, in which direction? Etc.)"

  Cullen Jennings wrote:

  "Why would you need 3265 - is this just for MWI"

  Rohan Mahy wrote:

  "Why is the SAS required to implement MWI? this is not motivated in the text. recommend deleting the entire line."

  Recommended Disposition: Remove the requirement for RFC 3265 from the specification.

  --------------------------------------------------------------------------------------------------------------------------------------------------------------------------

  RFC 3892 (The Session Initiation Protocol (SIP) Referred-By Mechanism) Support

  Ernst Horvath wrote:

  "I would like to see the following RFCs removed from the table in section 5: 3265, 3515, 3842, 3891, and 3892. The reason is that there are no use cases specified for these extensions in the rest of the document, and it is unreasonable to mandate global conformance to a generic mechanism like REFER or the event framework. (Which role is mandatory - subscriber, notifier or both - for which entity? Which methods can be referred, in which direction? Etc.)"

  Cullen Jennings wrote:

  "I might be wrong but I think that 3892 requires S/MIME - I doubt you want to do that "

  Rohan Mahy wrote:

  "This is also not motivated by the text. Referred-by strongly recommends the use of auth-id bodies signed with S/MIME. I am not aware of *any* implementations of this portion of the Referred-By spec. If we recommend Referred-By, we at least need to specify that this part is not required (and why)."

  Recommended Disposition: Remove the requirement for RFC 3892 from the specification

  --------------------------------------------------------------------------------------------------------------------------------------------------------------------------

  Call Progress Tones 

  Ernst Horvath wrote:

  "I have a real problem with section 15.1. The generation of local tones is an implementation matter and most likely subject to diverse 

  national regulations. Therefore this section should really be outside the scope of this specification."

  Cullen Jennings wrote:

  "The call progress tones are US only. This seems really bad for a sip forum document. I would be happy to contribute a list of other tones if you would like :-)"

  Rohan Mahy wrote:

  "The response code to tone mapping here is US centric and does not even accommodate variations among US phone systems (for example, some US phone systems use Fast Busy instead of SIT tone for "all circuits busy". Ideally the tones generated by the PBX will be configurable. I'm inclined to declare the specific tones out of scope."

  Recommended Disposition: Remove the specific call progress tone information and replace with a general statement that PBX systems SHOULD generate some form of call progress tone.

  --------------------------------------------------------------------------------------------------------------------------------------------------------------------------

  CDR Issues

  Hadriel Kaplan wrote:

  "The third paragraph [of section 9.1] confuses me greatly. I am not clear what CDRs have to do with interoperability. If the service provider chooses not to charge an Enterprise for anything, then they don't comply with this spec? (they won't stay in business for long, but that's another problem) It also implies a problem: how will the SAS know that a call truly came from an Enterprise's PBX, when it went through the service provider's SPS. It then lays out a solution for that by saying information identifying the Enterprise must be extracted from the cert and signaled to the SAS (somehow). That seems out of scope for an interop spec - if the service provider wishes to identify the enterprise PBX call through internally secured charging vectors, or call-ids combined with p-asserted-ids, or whatever, it's up to them. It won't impact interoperability unless you want the enterprise PBX to send some special header for this."

  Hadriel Kaplan wrote:

  "Bullet-4 of section 9:

  I'm pretty sure an interop spec should not be specifying what fields a service-provider should populate their CDRs with?"

  Ernst Horvath wrote:

  "Section 9.1, bullet 4. This seems to relate to the interface between Sip proxy server (on the SP side) and the SIP application server. I 

  believe this interface is outside the scope of this document."

  Recommended Disposition: Remove all references to population of CDR data from the specification.

  --------------------------------------------------------------------------------------------------------------------------------------------------------------------------

  Requirement for STUN Support on the Firewall

  Hadriel Kaplan wrote:

  "The diagram and text discuss a STUN server as an optional, non-mandatory element, yet the first paragraph of this section says the diagram outlines the minimal set of elements required to support the spec.  

  The diagram also has a dotted line to enterprise IP Phones, presumably for their use to learn and open pinholes for publicly routable media address/ports.  However the rest of the document does not distinguish between IP Phones and a PBX, but rather treats them as one logical system - in fact that's specifically called out in the definitions section for IP PBX and IP Phones.  Further, section 8 suggests service provider firewalls be STUN friendly, yet at no time does it say that service providers are recommended to use STUN, nor is that communication path in the diagram.

  In essence, it is simply one example method, as enum was in section 10 for number location.  It should be up to the Enterprise how they want to learn public addresses to use.  All you care about for interop purposes is that the addresses presented to the service provider and vice-versa are publicly reachable ones.

  Given that, and given that the SIP message details inherent in actually having IP Phones off an enterprise proxy/pbx are not really addressed in this document, I recommend you remove the STUN server from the diagram and section 5 standards list."

  Hadriel Kaplan wrote:

  "I still find it odd for an SP-to-Ent interop spec to tell an enterprise that they SHOULD do something on their firewalls, when that something is only 

  needed if they choose to fix something a certain way, and the way in which the fix it is up to them entirely and has no bearing on interoperability and is not defined in the spec itself.  

  It's like saying "your firewalls SHOULD support being a DHCP server", because after all the SIP PBX and phones need IP Addresses and DHCP is a way to get them, and some firewalls can be one. 

  Especially when you define the STUN server to be in the DMZ, whereas it could be implemented inside the firewall itself, or on the public side of a 

  firewall.  A DMZ is not on the public side of a firewall - and it may not even be public addresses.  I'm not clear how STUN would even work in a 

  traditional DMZ.  Sending packets from inside to a DMZ box, if even allowed, would not open any holes in the firewall to the public internet and would not provide you with the same public address/port for such."



  Janne Magnusson wrote:

  "A STUN friendly firewall must be a full cone NAT firewall and that are not common in enterprise environments as the security is much better 

  with a symmetric NAT firewall. 

  The STUN friendly firewall requirement is only valid if STUN is used, in other cases should the enterprise be free to choose firewall as they 

  like. In most cases will an enterprise want to keep the existing firewall and requirements like the suggested will hamper enterprises 

  willingness to implement SIP PBX technology. 

  I suggest that the last paragraph is removed. It is a pretty obvious requirement if you choose to implement a STUN solution that all network 

  elements must be STUN friendly."

  Recommended Disposition: Remove the requirement for the firewall to be STUN-friendly.

  --------------------------------------------------------------------------------------------------------------------------------------------------------------------------

  Requirement for Outbound Proxy Configuration on the SIP Application Server

  Ernst Horvath wrote:

  "In the 2nd paragraph of 6.1 and 3rd paragraph of 6.2, it is mandated that an FQDN is configurable for an outbound proxy. Why is it not left 

  as an internal matter how the proxy is addressed inside its own domain, e.g. by configuring its IP address?"

  Hadriel Kaplan wrote:

  "Like Ernst, I do not understand why an interop spec is mandating internal routing mechanisms in the service provider's network.  Calls that are routed for termination from the SP's SAS to its SPS may be done through numerous means, one of which may be FQDN-based.  It will not impact interop between the SP and Enterprise how this is done."

  Recommended Disposition: Remove the requirement from the specification.

  --------------------------------------------------------------------------------------------------------------------------------------------------------------------------

  Changes to the Firewall and NAT Traversal Section

  Rohan Mahy wrote:

  "propose we add a 4th item.  this was discussed very early, but was dropped.  this is consistent with our "do no harm" motto:

  4) SIP intermediaries MUST NOT modify IP addresses or port numbers in the body or Contact header of any message if any of the following are true:

  - The SIP message contains an "Identity" header (indicating use of the Identity extension)

  - the Content-Type of the body or any of its subparts is "application/pkcs7-mime" or "multipart/signed" (indicating support forSIP S/MIME)

  - any "application/sdp" body in the message contains any "a=candidate:" lines (indicating use of the ICE extension)

  - all the "c=" lines in any "application/sdp" bodies contain only public IP addresses (indicating that another element has already ensured the addresses are correct)"

  Hadriel Kaplan wrote:

  " Can you also add a vote for whether or not to add Rohan's addition to section 8?  I did not hear anyone else agree to it, and I strongly disagree to it. 



  There is no service provided/documented by this interop spec that requires this addition to be met, and there's a large sector of service providers and

  enterprises that can not and do not wish to meet it.  Normally I would not care if the doc said so because that sector of operators would simply ignore

  it, but for this particular requirement it would potentially break interoperability and suggest support for mechanisms which are not defined in

  this document.  

  It also references 2 IETF documents which are in draft form only - which I thought we were trying to avoid in this doc?"

  Recommended Disposition: NONE - A "yea" vote means keep the change, a "nay" vote means to strike the text in the final draft.








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