Martin Dolly presented a series of slides addressing each required feature
- voice services driven by regulations and customer demand
- certain services need to interop with PSTN/ISDN
- not all services are supported on all devices
   nor do all services interwork between technology domains
- Call Rejection
   - based on calling info not being presented to called party
   - new 4xx response code being defined for "anonymous call not allowed"
   - override of call rejection no longer required by TISPAN
     - however, we are looking at SAML assertions for these kinds of things
- Terminating Party Identification
   - provide authenticated identity back to calling party taking into
     account called party's identity restrictions
   - for purposes of future communication (callback)
   - provide Reverse Identity in 18x & 200 OK
     - 18x may be problematic since multiple UAs could respond
   - possibly a GRUU in response Contact
   - requirements seem to indicate the Identity should be for the
     "user" (AOR) not the "instance" (UA)
   - 3261 says display-name Anonymous, bad, should be something that has
     semantic meaning
   - new draft rosenberg-sip-identity-privacy proposes anonymous SIP URI
- Call Waiting
   - many saw this device requirement - nothing new required
   - however, the requirements seem to define this as a network based
     service where the network would determine that the user is busy
     - this is hard to know with SIP due to multiple devices
     - network may have a limit on the number of sessions user may have
     - network sends and indication to the called party that a call is
       queued
     - network also send indication to calling party that the call was
       queued
   - it was determined that we need to get some clarification of the requirements
- Call Diversion
   - use History-Info
   - needs resolution of service-retargeting
     - draft-jennings-sip-voicemail-uri-04
- Voice Capabilities
   - tied to ACR
   - CPC/OLI from PSTN
   - look at using SAML assertions (roles)
     - Rocky Wang & Hannes Tschofenig are writing a draft exploring SAML
- Malicious Call Identification
   - allow user to mark a call as malicious
   - e.g. interact with web server to log calling party
   - need to preserve anonymity of called party
   - Rosenberg-sip-identity-privacy has stuff that might help
   - only needs to flag the call to a server in the called party domain
     - e.g just Call-Id, From and To
   - ETSI uses PAI, not Identity
   - device has sufficient information to report the call
- Advice of Charge
   - what does this mean?
     - actual cost?
     - estimated cost?
     - id of call type?
     - upon call termination call start/stop
     - currency translation needed?
   - contact web server
(not mentioned in meeting, but why wouldn't this be an event package?)
- Call Completion on Busy/NoAnswer
   - presence or dialog package
   - there are implementation that interwork dialog package with TCAP
   - internet side gets complicated with multiple devices
     - really nasty
     - current implementation do a polling INVITE
     - dialog package is better
   - application server interacting with presence server
   - needs a design phase
   - do we need to support multiple devices per user?
   - need indication to calling party that service is available
- Next Steps
   - questions to clarify certain requirements
   - then what?
     - be proactive
     - investigate these features in a pure SIP world
     - let TISPAN propose solutions is same or different draft
   - Eric Sasaki & Martin Dolly will write a response "communiqué" draft

Miguel Garcia presented 1 slide
- Early Media Indication (not in the requirements document)
   - gate control
     - open gates during provisional response, fraud is possible
       - don't open forward gate until 200-OK
     - they propose a header in 1xx to indicate early media is authorized
     - the thing you are connected to is authorized to play early media