[SIPForum-techwg] IP PBX / SP Interoperability Draft Version 5

Chris Sibley chris.sibley at cbeyond.net
Fri Mar 10 12:31:39 EST 2006 

IP PBX / SP Interop WG team members,

Draft version five of the IP PBX / SP Interoperability specification is now
available online for your review at the following URL:

http://data.memberclicks.com/bbattach/128957/sf-draft-twg-IP_PBX_SP_Interop-
sibley-v5.pdf

This latest version of the draft incorporates the majority of the feedback
received since the last draft (see full release change notes below). 

Note that there are still a few minor items that require discussion (which I
will outline in an upcoming email), but I did not feel that any changes
resulting from those discussions would be significant enough to hold off on
releasing draft version 5 today.

As always thanks for your time and continued feedback, and good job to all!

--Chris

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

Draft Version 4 to (PREVIEW) Draft Version 5 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"


(PREVIEW) Draft Version 5 to (RELEASE) Draft Version 5 Changes:

Section 3, Reference Architecture

*	Removed the STUN server from the reference architecture drawing

	Section 4, Definitions

*	Updated the definition of the STUN server (changed to definition of
the STUN protocol)
*	Added definition for the TURN protocol
*	Added definition for the ICE mechanism


	Section 7.2, Service Provider Requirements

*	Added wording to indicate that registrations are not required to be
sent by the PBX

	Section 9, Firewall and NAT Traversal

*	2nd paragraph: changed 'i.e.' to 'e.g.'
*	Removed the requirement for the firewall to be STUN-friendly
*	Removed first two requirements related to SIP intermediaries

	Section 6, Standards Support

*	Removed requirement for E.164 support from SPS
*	Removed all references to RFC 3265 support
*	Removed all references to RFC 3515 support
*	Removed all references to RFC 3824 support
*	Removed all references to RFC 3842 support
*	Removed all references to RFC 3891 support
*	Removed all references to RFC 3892 support
*	Removed all references to RFC 3959 and 3960 support

	Section 10.1, Authentication of the Enterprise by the Service
Provider

*	Removed all references to population of CDR data

	Section 16, PSTN Interactions

*	Removed the specific call progress tone information and replaced
with a general statement that PBX systems SHOULD generate some form of call
progress tone
*	Removed section 15.2 and all references to RFC 3398 support
*	Removed all references to RFC 3959 and 3960 support

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