[SIPForum-techwg] IP PBX / SP Interoperability Proposed Final Draft

[Chris Sibley]

IP PBX / SP Interop WG team members,

 

The proposed final draft of the IP PBX / SP Interoperability
specification is now available online for your review at the following
URL:

 

http://data.memberclicks.com/bbattach/130808/sf-draft-twg-IP_PBX_SP_Inte
rop-sibley-final.pdf

 

This version of the draft incorporates the last bit of feedback received
from the 3/10 - 3/22 comment period. Full change release notes can be
found below.

 

Note that the only outstanding item I have documented is the proposed
terminology change from "SIP Application Server" to "SIP Gateway
Service". The final version of the draft still refers to the device in
question as "SIP Application Server" since I received only one comment
supporting the change (Rohan Mahy) and one against the change (Sofia
Nekrasovskaia).

 

Finally, I wanted to express my sincerest gratitude and compliments to
all of the team members / contributors. IMHO, your extensive technical
knowledge, professionalism, and attention to detail are definitely
reflected in the final product! It has genuinely been a pleasure working
with all of you!

 

Best Regards,

 

--Chris

 

Draft Version 5 to Final Draft Changes: 

 

Section 8, Signaling Security

 

*	s/Validation steps include checking the status of the
certificate as well as the status of all the certificates in the
certificate chain using CRLs or other mechanisms such as OCSP/Validation
steps include checking the status of the certificate as well as the
status of all the certificates in the certificate chain using CRLs or
other mechanisms such as OCSP

 

Section 10, Authentication and Accounting

 

*	Broke into two sections (10.1.1 and 10.1.2) for better
organization. Added text indicating that SAS must support both methods
and the IP PBX must support option 1 at a minimum.

 

Section 12.3, 'To:' Field - Emergency Services Destinations

 

*	Added ;user=phone tag to URI examples.

 

Section 15.1, Media Capability Negotiation

 

*	Added requirement for any device that terminates RTP traffic to
include the directionality attribute for all generated SDP
offer/answers.

 

Section 15.3, Transport of DTMF Tones

 

*	Removed requirement for DTMF mode to be user configurable.

 

Section 16.2, Early Media

 

*	Added requirement for PBX to cut-through media and disable
locally generated call progress tones when media is received on any
previously offered recvonly or sendrecv media stream.

 

 

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