[SIPForum-techwg] IP PBX / SP Interop Draft Version 5 "preview" and next steps...

Chris Sibley chris.sibley at cbeyond.net
Fri Mar 3 11:47:03 EST 2006 

IP PBX / SP Interop WG team members,

I have completed making all discussed changes from the 1/27 - 2/26 comment
period. I have also completed making many of the smaller changes requested
from the 2/27 - 3/2 comment period (the full list of changes made can be
found at the end of this message). 

A "preview" of the current working text is available for your review at the
following URL:

http://data.memberclicks.com/bbattach/128012/preview--sf-draft-twg-IP_PBX_SP
_Interop-sibley-v5--preview.pdf

As I indicated yesterday, I have compiled a "master list" of contributor
comments from the 2/27 - 3/2 time period. From this list, I have extracted
the set of issues that more than one person commented on.

As a next step I would like to see if we can get to a quick consensus on
these "top" issues so that I can go ahead and make the necessary changes to
the working draft. 

I will be sending out a separate email in a bit that contains this list of
issues, the applicable comments made by each of the contributors, and my
recommendation for how to proceed on each. If you would, please take the
time to give the recommendation for each issue a quick "yea or nay" vote so
that I can quickly gauge the group's overall consensus on these particular
issues.

Finally, after this voting round is done, I'll send out a consolidated list
of the remaining items from the "master list" that still require discussion
and we can try to hammer those issues out.

Thanks!

--Chris

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

Draft Version 5 (preview release) changes


Section 1, Introduction

*	Added statement to clarify that the primary service intended for use
by the spec is audio-based PSTN call origination/termination
*	Added references to ITU-T Recommendations in addition to IETF RFCs
*	Added following statement to 3rd paragraph: "Note that this document
does not preclude or discourage the negotiation of additional
functionality."
*	3rd paragraph: s/This SIP Forum recommendation is proposed to help
address the issue./This SIP Forum document aims to address this issue./ 
*	3rd paragraph: s/for predictable interoperability between/for a
predictable interoperable scenario between/ 

Section 3, Definitions

*	Updated reference architecture drawing to show use of STUN between
PBX and STUN server (in addition to IP phones)
*	Broke out the reference architecture drawing and the definitions
into separate sections
*	Added definition for Application Layer Gateway
*	s/the minimal/common/ 

Section 4, Key Assumptions and Limitations of Scope

*	Added statement to clarify that the primary service intended for use
by the spec is audio-based PSTN call origination/termination
*	Modified item #6: s/911 issues/calling issues, for example routing
to national emergency numbers such as 911, 112, 999, or 000, / 

Section 5, Standards Support

*	Added RFC 2782 as MANADATORY requirement for the SPS
*	Changed column title 'RFC #' to 'Standard ID' 
*	Legend: s/(Receive only)/(at minimum to Receive)/ 

Section 6.1, (Locating SIP Servers) Enterprise Requirements

*	Spelled out FQDN.
*	4th paragraph: changed AOR references to URI
*	Changed wording in last paragraph from "register one or more AORs"
to "register a contact URI against one or more AORs".
*	s/operate/ensure the existence of/ 

Section 6.2, (Locating SIP Servers) Service Provider Requirements

*	s/to update the SAS's default IP address for the PBX/to update a DNS
entry associated with the PBX in a DNS zone managed by the Service
Provider./ 

	Section 7, Signaling Security

*	Updated text to require the use of canonical hostnames by the SPS in
the 'Via:' and/or 'Route:' SIP header fields.
*	5th paragraph: s/matches the server's host name or SIP URI/matches
the host portion of the target URI/ 

Section 8, Firewall and NAT Traversal

*	Removed requirement that symmetric NATs must not be in the
communications, and that SIP ALGs should be disabled.
*	Modified text to reflect the fact that the specification does not
exclude the use of any particular NAT traversal method (e.g. static config,
SBC, SIP-aware firewall.)
*	Added clarification that IP addresses contained with SIP message
bodies as well as headers must be publicly routable addresses.
*	Added requirements for SIP Intermediaries

Section 9.1, Authentication of the Enterprise by the Service Provider

*	Minor text change ("The username supplied..." -> "The authentication
username supplied...")
*	Added requirement to challenge the REGISTER request (in addition to
INVITES)

Section 10, Enterprise PSTN Identities

*	Changed references of Address of Record / AOR to URI.

	Section 11, Enterprise URI Formatting and Addressing Rules

*	Added statement that both open and closed numbering plans must be
supported.
*	Added guidance for formatting the Request-URI field 

Section 11.1, (Enterprise URI Formatting and Addressing Rules) 'From:' Field

*	Clarified the text to indicate that Option 1 is the preferred
method.

	Section 11.2, (Enterprise URI Formatting and Addressing Rules) 'To:'
Field - PSTN Destinations

*	Corrected URI examples in 11.2.1 and 11.2.2 (added international
prefix symbol and ;user=phone tag where applicable)

	Section 11.3, (Enterprise URI Formatting and Addressing Rules) 'To:'
Field - Emergency Services Destinations

*	 s/ALI/emergency location information/ 

	Section 12, Service Provider URI Formatting and Addressing Rules

*	Added guidance for formatting the Request-URI field 

	Section 12.1, (Service Provider URI Formatting and Addressing Rules)
'From:' Field

*	Added second option for anonymous URI format (to make compatible
with Identity: header)

	Section 12.1.1, (Service Provider URI Formatting and Addressing
Rules) ('From:' Field)  Option 1: Utilizing the 'From:' and
'P-Asserted-Identity:' SIP Header Fields

*	3rd paragraph: s/"Trust Domain"/"Trust Domain", as defined in RFC
3325/ 

	Section 13, Quality of Service Considerations

*	Removed unnecessary statement indicating that the SP and Enterprise
must agree on settings if the recommended values are not used.

	Section 14.2, CODEC Support and Media Transport

*	2nd paragraph: s/terminates RTP traffic MUST/terminates RTP traffic
over UDP MUST/ 

	Section 14.3, Transport of DTMF Tones

*	Clarified that TGs must support RFC 2833 with any codec

	Section 14.4, Echo Cancellation

*	Clarified that some of the requirements only apply to devices that
supports fax and/or modem transmissions.

	Section 14.5, Fax and Modem Calls

*	Clarified that some of the requirements only apply to devices that
supports fax and/or modem transmissions.
*	2nd paragraph: s/SIP RE-INVITE method/SIP reINVITE request/ 
*	2nd paragraph, appended to end: " or SIP UPDATE request as described
in RFC 3311 [13]."

Section 16, References

*	Added ITU-T Recommendation T.38 to the list of references and tagged
the text as applicable

	Section 17, Contributors and Contact Information

*	Updated contributor contact information
*	Changed 'Email:' to 'mailto:' in contributor contact information
section

Miscellaneous

*	Minor wording / grammar fixes
*	Replaced all occurrences of "calling name information" with "display
name information"
*	Replaced all occurrences of AOR and address of record with "URI"
*	Changed all references of "CODEC" to "codec"


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