SIPPING Session 1, Monday March 17th, 2003 19:30-22:00

Reported by Vijay Gurbani and Mary Barnes


Minor change from posted agenda; ieprep stuff moved to Wed. afternoon. Gonzalo to talk about Early Media

Alan Johnston, Services Example

SIP Service Example I-D: Recent changes: new flow showing click2dial using REFER.  Replaced domain names w/ examples.com.
To Do: actual flows really stable except couple of them; need to add text discussing how to implement services and tradeoffs between them.  By next time, this I-D should be complete.

Transfer: This version had few changes: added a section on use of Referred-By header.  Added selected message details.  Added many more call flows on attended transfer and unattended transfer.  Added a security consideration section.  No known open issues with the I-D. Open for WGLC.

Gonzalo: about a globally routbale URI, JOnathan is presenting something. Will it have any influence on your I-D. Alan: No.  But if we agree on a solution it will be nice to reference it.

Rohan: We have 3 RFCs which have been published, considering the amount of work we had on our plate.  2 on IESG queue; bunch of individual I-D. 3 in here that are ready for WGLC.  A lot of work is blocked on other stuff -- call control, ICE, work going on in midcom, etc.

Node about publishing individual docs: it does not mean we do an expert review.  We do it only when we have a P header.  A couple of folks have asked for an expert review; we may do an announcement to the SIPPING WG list and announce that FYI some work may be beneficial to the community in general and may benefit from a review.  Such documents should not affect WG documents.  

Maybe useful for us to split things into smaller groups -- some I-Ds document usage of SIP, others talk about services (cc-conferening, cc-transfer).  It may be appropriate to let some of this work proceed without supervision of the WG.  We may end up doing this in Vienna in order to get up to speed.  Will like to get comments on the mailing list or after session on if this is okay.

Jonathan Rosenberg, Caller Preferences Use Cases.

Problem statement: not clear what problems caller-prefs is trying to solve.  This I-D tries to make this more obvious by defining use cases.  Will provide some explanatory text on how to do this in a proxy.  Propsal: adopt as a work item for an informational RFC. <Hum level taken; adopted as WG item>

ICE, Jonathan Rosenberg

We don't have a sufficiently good answer for NAT traversal in SIP. Depends on business models, many different ways to solve it.  10 different solutions is as good as no solution.  STUN, TURN, MIDCOM, RSIP, etc. are all piecemeal solutions.  How can we expect interop? ICE is not a new protocol; it is a methodology for using address fixing protocols.  ICE explains how to use the other protocols for NAT traversal.  ICE properties:
    - Always will find a means for communicating if one physically exists.
    - Always find the lowest latency communication path.
    - Always finds the communication path cheapest for the service provider.
    - Does not require any knowledge of topology, NAT types or anything.

I-D is not trivial; this is not a trivial problem.  Benefit is that it is one algorithm which lives in a client.
Henning: will mobility introduce some problems?
Jonathan: Not if IP address changes outside of 1RTT.
Drawbacks: both sides have to support it.  May require an extra RTT in call setup.  Will reveal private IP addresses to outside world. Requires muxing STUN and media on the same port.

Proposed plan: solves industry problem - lack of cohesive way to work across NATs now.  Also facilitates IPv4->IPv6 (and vice versa) inter-work cases.  Pre-conditions can be specified so that phones are never rung unless both endpoints can hear each other.

??: This may be relatively heavy protocol.  If we can convince everyone to support this, it is okay.  Otherwise it is hard to work it in practice. Jonathan: This is a hard problem; how do we make this hodge-podge work? ??: This just describes the hodge-podge. Jonathan: How do I determine, if I am a phone vendor, which scenarios to support? ??: I am sceptical that it will work -- relatively heavy for client implementations.  

Juha: There are NAT implementations that understand SIP.  Why not work on these implementations instead of having clients to change?  We should not give a message to the NAT vendors that they do not have to do anything. Jonathan: It is easier to upgrade a pair of phones than hoping that the NATs will work.

Rohan: How many people will implement this?  <A couple of hands raised>.
Will one of the implementor write a short I-D on the pre-conditions in  SIP.

Dave Oran: Even if you do this in endpoints, other things in the middle will break.  Expectation of what pepole have today is at least somewhat correlated with reality.  My concern is that this may raise expectations.

Rohan: Hum on if we want to pursue this and document what happens when you fall back.  Hum on: If you think you will like this to be documented by the sipping
WG and also to have the fallbacks mechanism to be documented by the sipping WG.  <Hum succeeded>

Jonathan Rosenberg, Session Policy Requirements.

What happens when a service provider wants to insist on some aspects of a session (media flows through a realy, video not be used, etc.) So far this has been done with SDP modification.  Well known problems: does not work with data encryption, requires proxies to have knowledge of session descriptions, proxy complexity increases, ...

Solution: 3GPP does this -- they use 488 as a response code to list codecs which are acceptable.  This has some drawbacks as well.

Do we want to work on this problem?  That is the real issue for us to decide now.

Many people approached the mic and agreed to the requirements being valid.

Gonzalo took a hum level on taking the requirements as base for the work going forward.  Hum level indicated interest.

Jonathan Rosenberg, URI Leasing.

Problem: SIP requires a call to be routed to a specific UA instance. Would also like an alternative to dialog sharing.  

Globally ROutable UA URI (GRUU): routable from anywhere, routes to a specific UA instance, and maybe temporaily scoped (for cases like conference I do not want the URI to be valid forever; after the conference ends, the URI should become invalid).

Alternatives for getting a GRUU: Use user AoR; use device IP as URI (NAT, firewall problem); caller preferences; embeded route headers (not allowed in Contact URI, long term loose Routes have reliability problems; route may only work from certain places).

Gonzalo: Need more discussion and work on mailing list before reaching  any conclusion right now.

App Interaction Design team, Jonathan Rosenberg

Some discussion on KPML: SIP vs. HTTP issue on posting requests. Multiple apps: one app consuming input destined for other app. Continuing issues: framework scope and complexity, supporting immediate barge functionality in KPML, lifecycle management of UI components.

Dean: What if I want to use KPML to map my keyboard to a conference control application.  Can I do that with KPML?

Jonathan: Please send the use cases to me so we can document them for consideration into a possible use cases document.

Alan Johnston, SIPPING Conferencing Design Team.

We have 6 docs that the design team has put out so far; some more forthcoming soon.  High-level reqs I-D; 2) a framework I-D; 3) call control for conferencing I-D (add users) The following I-Ds relate to conference policy -- how do we manipulate the media for desired mixing, how to setup a conference, access control, ...1) ...
2) Media policy data reqs I-D; how media is mixed and how mixing ops are performed 3) Media policy manipulation - looks at media operations to change the topology (e.g. creating a side bar).

Related documents: conference package, floor control reqs, floor control using SOAP, and caller prefs.
 
Status of work: reqs, framework and call control are relatively stable; should fit together and tell a consistent story.  Ready for adoption by WG as official documents.

Other I-Ds relate to conferenc policy -- split into two parts: memberships and general policies -- stable.  Media policies is in infancy.  Need feedback from the group on media work.

Policy group needs working group home.  Rohan: mmusic will NOT be chartered to do this; so it is either us or a new BOF.  Will work it out with the ADs later.  We have NOT (ed: corrected by dwillis) been told that we have to stop working on this.

Open issues: conference policy protocol, floor control, notifications.

Dean: Reqs, framework, and call control I-Ds are ready to be approved  as WG documents.  Proposal: we adopt them as WG docs. Dean called for a hum; proposal accepted.

Open reqs: discovery, focus role migration, cascading and camera control.

Gonzalo Camarillo, Media encoding for hearing impaired users Status: have not progressed much since last IETF.  Now that we have media policy I-D and session policy reqs, we will make more progress for the next IETF.

Jon Peterson, SIP Emergency

<Trying to backfill previous notes; Jon's talk was too short to catch up.  Hope someone else was able to capture Jon's notes>.

Event Throttle Requirements, Aki Niemi

Aki went through the model and the requirements (see slides). Issues: Is the model accurate and appropriate? Do we need both leaky-bucket and the strict throttle models, or is only one of them enough?

Dave Oran: Quarantining and throttling are not the same thing. One introduces delay the other control rates.

Rohan: Jonathan, Dave Oran and Dean will review the I-D and the  update from them before this becomes a WG document.


Persistent Connection Management Requirements, Vijay K. Gurbani

   draft-jain-sipping-persistent-conn-reqs-00.txt
   draft-ietf-sipping-connect-reuse-reqs-00.txt

Background:
Based upon Rohan's original proposal

This proposal: one bi-directional persistent connection model:
Pros:

Cons:

2 solution options:

Discussion:

 
Conclusion:

Continuing discussion:

Continuing discussion: