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Date: Wed, 30 Apr 2003 23:24:03 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [Sipping] Conferencing Framework open issue: scope of data which is manipulated
 by CPCP
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Eric raised an issue in his comments on conf-fw, to which I replied:

>> Also, w.r.t. section 5.8.2, I would really hope that CPCP is SIP:
>> I'm really hesitant to have to implement all the same protocol
>> actions with two different protocols "just because."  Let's pick
>> one and run with it.
> 
> 
> See above discussion on why these are just not the same. In the case
> of conference participants, you will basically learn them as the
> result of a full-state update generated by a subscription to the
> conference state.  CPCP, being a data transaction protocol, would
> also allow you to directly query for the list. Just because there is
> this sliver of overlap does not mean that they are the same protocol.
> 

My response is wrong. In the current design, you cannot use CPCP to 
access any of the focus state, such as the current participant list. It 
is strictly viewed as a protocol for manipulated the conference POLICY.

This leads to the question - is this sensible? On one hand, one would 
expect that a protocol for accessing state of the conference should give 
you access to ALL of the state, not just the subset which is defined as 
policy. On the other hand, as Eric points out, conference policy and 
conference state are very different. Indeed, they may likely reside in 
entirely different servers in the network. One can imagine using an 
external DB to hold the policy, whereas focus state is in-memory 
soft-state. Thus, using the same protocol to access them may be 
complicated to support.

What do people think about this issue?

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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Subject: Re: [Sipping] Thoughts on GRUUs, URI leasing and embedded route headers
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inline.

Christer Holmberg wrote:
> Hi,
> 
> 
>>>To make this work, it is assumed that requests sent in the backward
>>>direction really will be directed via the proxy which issued the
>>>leased URI (unless the proxy is able to "share" the information with
>>>other proxies). So, won't we have the same problem with the
>>>Record-Route URI that you now try to solve for the Contact URI - ie
>>>we want to make sure that requests reach a specific node, and not
>>>just any node within a specific user domain?
>>
>>No.
>>
>>Just because I ask a specific server for a URI that is routable in the
>>domain, does not meant that requests have to go to that specific server.
>>  It is a requirement on the administrator of the domain to guarantee
>>that the URI will route to the UA instance when used by anyone from
>>anywhere. This will frequently require coordinated routing policies
>>throughout the domain. Thats fine. Thats why you need to ask for the
>>name - to make sure what you get is consistent with those routing policies.
> 
> 
> Ok, but assume that proxy A still wants a request to be sent via it. Now, it
> can do so by inserting its IP address in the Record-Route URI, but aren't the
> problems them the same as if the UA would insert its IP address in the Contact?

The problem GRUUs are trying to solve is that a UA wants a URI that it 
can be reached at *outside of the dialog* that was just set up. I do not 
see a pressing need for a proxy to be reached *outside of a dialog* that 
was just set up.

> 
> My question basically is: should it only be possible for UAs to obtain a GRUU,
> or should it also be possible for proxies?

Its possible but I see no value.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
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Use sip@ietf.org for new developments of core SIP



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Sorry for not responding to these questions earlier. Some answers inline:


Arjun Roychowdhury wrote:
> Reading 
> http://www.ietf.org/internet-drafts/draft-levin-sipping-conferencing-requirements-03.txt
>  and 
> http://www.ietf.org/internet-drafts/draft-koskelainen-sipping-conf-policy-req-00.txt
>  in conjunction,
> 
> it seems both CPCP and SIP have similar requirements and the reqts
> listed in the above two drafts to a great extent can mix with each
> other. This probably is a good  thing since that means SIP can meet
> many of CPCP's requirements.
> 
> But it does not clarify my confusion below:
> 
> In addition, framework-01, for each scenario lists a SIP way and a
> CPCP wa
> 
> Even though SIP may finally land up doing a lot of what is required
> by CPCP,  looking at the framework-01 diagram Figure 2, the interface
> between Participant and Focus is different from the interface between
> Participant and Conference Policy server.
> 
> However, when I read petri's, jonathan's and orit's drafts in
> conjunction, I see these two interfaces sparring in the same ring.
> 
> I am somewhat confused. Where does the design team consider CPCP to
> stop (Participant - CPS) and SIP dialog to start (Participant to
> Focus) ?

Firstly, we arent defining these protocols based on the fact that they 
go from element A to element B. We define them based on their functional 
properties.

In my mind, the difference between SIP and CPCP is clear. SIP is about 
initiating and managing sessions. CPCP is an RPC type of protocol that 
manipulates policies on a server. The confusion is arising because 
somtimes the establishment of a policy results in the creation of a 
session, or vice-a-versa, so that the establishment of a session is 
effectively creating some kind of change in policy.

 From the full scope of CPCP, though, its clear that it is doing many 
things that make no sense for SIP. For example, setting up policy about 
who is allowed to moderate, or setting policy about who can and can't 
join the conference. These things are clearly not SIP.

Thanks,
Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
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Marjou Xavier wrote:
> Hello,
> 
> Something is not clear to me : is CPCP really mandatory to perform
> something as "simple" as calling many recipients via a focus (adhoc
> conference).

In an ad-hoc conference, the focus resides with one of the endpoints. 
Therefore, an internal interface can be used to instruct the focus to 
call out to additional users. So, in the case of ad-hoc, something as 
"simple" as mass invitation does not require CPCP.

With a centralized conference server, a client can use a series of REFER 
requests to bring new users into the conference. This is described in 
the document as a SIP mecahnism for adding users. However, for a large 
number of invitations, this is really inefficient, and thus a real 
control protocol like CPCP is needed.


> In my opinion, it should be as simple as sending an
> e-mail to multiple recipients (Of course, having everybody in the
> "To" header is not possible in SIP, but having another header for
> giving the list of recipients whould be very nice).

Its not as simple as that. How would you report the results of that back 
to the sender?

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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Date: Wed, 30 Apr 2003 23:43:09 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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I was combing back through the SIP/SIPPING list to make sure all 
conferencing framework comments and issues were addressed, and I noticed 
that there was no response to this.

The design team considered whether or not to consider conferences that 
were split amongst multiple focuses, where the focuses were actively 
communicating in order to make the whole thing appear as a single 
conference. This is a very hard problem, and given the scope of the work 
already on the table, we ruled it out of scope.

You can have one focus be a member of another conference. That requires 
no standardization, since a focus is just a UA. That seemed to get us 
enough functionality for free, that the more complex case wasnt worth 
the cost.

-Jonathan R.


Zon-Yin Shae wrote:
> 
> 
> 
> Overall the draft looks good.  I wish this conference frame work can be
> extended to cover the following enterprise/service conferencing scenario:
> 
> When an application need a conference service, it only creates a conference
> ID (and its associated policy).  By the time the application does not know
> (does not want to or can not specify) which conference service ( Focus) to
> use.   Moreover,  it is desired for the performance and scalability that a
> set of Focuses should work collaboratively to perform the distributed
> conference service.  For example, 20 participants in a conference,   10 in
> CA and 10 in NY,  it is better to have two Focuses (one in CA and one in
> NY) to jointly perform the service.   Protocols among Focuses are required.
> 
> Also, most of VoIP phone does support ad-hoc conference capability and is
> an useful media mixing resource.  It can be useful (for the resource usage
> and scalability) that the Focus can delegate some of the media mixing
> function to be performed in these phones of the conference participants.
> Protocols among Focus and participants are required.
> 
> There are some components are required in the above scenario to achieve the
> dynamic conference configuration.     .
> 
> 1.  A conference is identified by a conference ID.  A conference might be
> collaboratively serviced by more than one Focuses.  Since URI is used for
> routing to only one UA, it would be more appropriate to carry the
> conference ID (and its associated policy) in the payload with standard URI.
> 
> 2. The default Focus for a UA can be pre-configured or discovered using
> similar procedure of outbound proxy (or proxy server).
> 
> 3. Protocols among Focuses (and conference policy server) are required.
> The Subscribe and Notify SIP mechanism with extended SDP can be used among
> Focuses to establish a collaboratively distributed service for a
> conference.
> 
> 4. Protocols between Focus (and conference policy server) and participants
> are required.   In Figure 2, the conference policy control protocol is only
> one way (from participant to conference policy server), it can be two way
> protocol to make use of the mixing resources in the participants (e.g.,
> VoIP phone).   One possibility is to extend REFER mechanism using
> (extended) SDP for carrying conferencing policy.
> 
> Regards,
> 
> zon-yin shae
> IBM Watson Research Center
> 
> 
> aroychow@hss.hns.com@ietf.org on 02/22/2003 10:37:44 AM
> 
> Sent by:    sipping-admin@ietf.org
> 
> 
> To:    <shaj@cisco.com>
> cc:    sipping@ietf.org, sipping-admin@ietf.org
> Subject:    Re: [Sipping] Comments on
>        draft-rosenberg-sipping-conferencing-framework-01.txt
> 
> 
> 
> 
> Inline prefixed by <ARC>
> 
> The idea of having a unique URI for each conference is a good one. However
> a more scalable approach would be to have a master Focus or a group of them
> that register with a standard URI. This way users use a standard URI with a
> Conference ID as the payload in the Invite message.  This is directed to
> the master Focus that does the following
> * Retrieve the Conference ID
> * Check if a Focus corresponding to the conference exists
> * If the Conference specific Focus  does not exist create a new Focus with
> a unique URI derived from the Conference ID
> * Forward the call using the standard Refer mechanism to the new Focus with
> the Unique URI.
> 
> <ARC>
>  You may also want to read
> http://www.iptel.org/info/players/ietf/conferencing/draft-johnston-sipping-cc-conferencing-01.txt
> .
> Its provides the definition of a conference Factory URI, which could be
> used to create new conferences and their Focii.
> Note that when you send an INVITE to a factory URI, that entity returns the
> actual Conference-URI in the Contact header.
> Also, you may be using a central 'conference enabling application'
> that  asks you lots of questions before allowing you to create/join  a new
> conference. See an example of that in 4.3 of alan/orit's draft.
> Note however , that when you want to use a "standard wellknown URI" that
> has not been published to you by the conference server,
> you are assuming that there is some agreement for that domain on conference
> naming policies.
> I am not sure I like your idea of a payload contaning conf-id (what use is
> that ?)
> Also, note that for scalability (if thats your primary concern), this
> mechanism allows a way for you to start with a 'guessed URI' and from
> then on communicate directly with you assigned focus without routing
> everthing to what you call your 'master focus'.
> </ARC>
> 
> 
> Instead of dealing with Conference aware UA's and Conference Unaware UA's
> we could assume all participants in the Conference to be Conference Aware.
> The Conference Unaware participants are routed through a Gateway (e.g. an
> IVR ) that provides a primitive interface to the conference unaware
> participants and a Conference aware interface to the Conference
> itself.  This would keep the model simpler and it can allow Conference
> aware UAs a way to control the conference.
> 
> <ARC>
>  It is extremely important to acknowledge conference-unaware endpoints. You
> cannot assume an IVR or any other application in all
> conference scenarios who 'aware' on behalf of an unaware UA. Again, reading
> both Jonathan and Alan/Orit's draft in conjunction may motivate
> you more on the requirement of conference unaware UAs. Remeber also , that
> most of the EPs deployed in SIP have zilch idea of what isFocus is
> as well as will not be aware of protocols to moderate a conference for a
> while.
> </ARC>
> 
> 
> The other comment was in the area of the Conference policy control
> protocol. Is this something like the SDP which is carried as a payload in
> SIP or does it have specifics of it's own. Currently most of the operations
> that are part of the CPCP are also possible through SIP based mechanisms.
> Is there a need to define two parallel mechanisms within the same
> framework.
> 
> <ARC>
> I think what has been identifed here is a need for a conference policy
> control protocol. It is not ascertained yet which protocol will deliver
> these
> requirements. Thats as far as I know. The existing SDP specs do not meet
> all control requirements that come to my mind.
> 
> Alan - would you mind reposting that link of conferencing work in
> progress  ? I thought I noticed some drafts attempting to address this in
> more detail
> 
> Now a question again for the authors
> of  draft-johnston-sipping-cc-conferencing-01.txt:
> 
> * section 4.5 callflow - you show alice sending an invite to a conf-factory
> URI but the ladder diagram shows it being received by a 'Focus'.
> The definition of a focus
> 
> 
> 3.1 Focus UA
> 
>   A focus, as defined in the framework, hosts a SIP conference and
>   maintains a SIP signaling relationship with each participant in the
>   conference. A focus contains a conference-aware user agent that
>   supports the conferencing call control conventions as defined in this
>   document.
> 
> Since a 'Focus' hosts 'a specific' conference, I find it a little
> disconcerting
> that the call flows show messages with a factory-URI reaching a focus. I
> thought
> it may be clearer, if we limit a 'focus' to be the target of a specific URI
> meant
> for a specific conference. If a person sends an INVITE to a factory-URI or
> a 'conf-app-URI' < like an IVR etc which then contacts a Factory after auth
> etc>
> it should not be received by a Focus, should it ?
> 
> </ARC>
> 
> regds
> Arjun
> 
> --
> Arjun Roychowdhury @ Hughes Software Systems
> 11717 Exploration Lane, Germantown MD 20876
> (O): 301 212 7860  (M): 240 997 0066{@vtext.com}
> 
> 
>                                                                            
>      "Shiva"                                                               
>      <shaj@cisco.com>                    To:        <sipping@ietf.org>     
>      Sent by:                            cc:                               
>      sipping-admin@ietf.o                Subject:        [Sipping]         
>      rg                          Comments on                               
>                                  draft-rosenberg-sipping-conferencing-fram 
>      02/22/2003 01:57 AM         ework-01.txt                              
>      Please respond to                                                     
>      shaj                                                                  
>                                                                            
>                                                                            
> 
> 
> 
> 
> Hello,
> 
> Overall the draft looks good.  Here are some of my comments on this draft
> 
> 
> The advantage of doing this is that I could have multiple Conference
> servers that are running that are registered their master focus.
> 
> We can implement Load Balancing across Conference servers without the UA
> realizing it as it is not connecting to a specific Focus on a specific
> server. A second advantage is that if in the course of a meeting if the
> focus were to go down or the server went down the participants can attempt
> to rejoin the call. In such a case the Master Focus would see that the
> Conference specific focus is down and recreate it allowing users to quickly
> rejoin the conference.
> 
> Of course one needs to work out the details of the communication and
> keepalive between the Master Focus and the Conference focus it creates.
> 
> The other advantage is if there are global Conference policies these can be
> applied at the master and the Individual Conference focus needs to worry
> about the specifics of the conference it is dealing with.
> 
> The other comment was in the area of the Conference policy control
> protocol. Is this something like the SDP which is carried as a payload in
> SIP or does it have specifics of it's own. Currently most of the operations
> that are part of the CPCP are also possible through SIP based mechanisms.
> Is there a need to define two parallel mechanisms within the same
> framework.
> 
> 
> Instead of dealing with Conference aware UA's and Conference Unaware UA's
> we could assume all participants in the Conference to be Conference Aware.
> The Conference Unaware participants are routed through a Gateway (e.g. an
> IVR ) that provides a primitive interface to the conference unaware
> participants and a Conference aware interface to the Conference
> itself.  This would keep the model simpler and it can allow Conference
> aware UAs a way to control the conference.
> 
> Regards
> Shiva
> Cisco Systems
> 
> 
> 
> 
> 
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
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This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        sipping
	 <sipping@ietf.org>
Subject: RE: [Sipping] Conferencing Framework open issue: scope of data wh
	ich is manipulated by CPCP
Date: Thu, 1 May 2003 09:33:04 -0400 
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Generally I agree that state and policy are separate.
OTOH, I think the vast majority of implementations will have
them in the same implementation.  I think the design should
allow the possibility of the functions being split, but
be optimized when they are the same entity.  That means they
should have nearly 100% overlap on representation of data
and protocol machinery.  

Brian

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Wednesday, April 30, 2003 11:24 PM
> To: sipping
> Subject: [Sipping] Conferencing Framework open issue: scope of data
> which is manipulated by CPCP
> 
> 
> Eric raised an issue in his comments on conf-fw, to which I replied:
> 
> >> Also, w.r.t. section 5.8.2, I would really hope that CPCP is SIP:
> >> I'm really hesitant to have to implement all the same protocol
> >> actions with two different protocols "just because."  Let's pick
> >> one and run with it.
> > 
> > 
> > See above discussion on why these are just not the same. In the case
> > of conference participants, you will basically learn them as the
> > result of a full-state update generated by a subscription to the
> > conference state.  CPCP, being a data transaction protocol, would
> > also allow you to directly query for the list. Just because there is
> > this sliver of overlap does not mean that they are the same 
> protocol.
> > 
> 
> My response is wrong. In the current design, you cannot use CPCP to 
> access any of the focus state, such as the current 
> participant list. It 
> is strictly viewed as a protocol for manipulated the 
> conference POLICY.
> 
> This leads to the question - is this sensible? On one hand, one would 
> expect that a protocol for accessing state of the conference 
> should give 
> you access to ALL of the state, not just the subset which is 
> defined as 
> policy. On the other hand, as Eric points out, conference policy and 
> conference state are very different. Indeed, they may likely 
> reside in 
> entirely different servers in the network. One can imagine using an 
> external DB to hold the policy, whereas focus state is in-memory 
> soft-state. Thus, using the same protocol to access them may be 
> complicated to support.
> 
> What do people think about this issue?
> 
> Thanks,
> Jonathan R.
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Scientist                             Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> 
> _______________________________________________
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.

	Title		: A Framework for Conferencing with the Session 
                          Initiation Protocol
	Author(s)	: J. Rosenberg
	Filename	: draft-ietf-sipping-conferencing-framework-00.txt
	Pages		: 39
	Date		: 2003-5-1
	
The Session Initiation Protocol (SIP) supports the initiation,
modification, and termination of media sessions between user agents.
These sessions are managed by SIP dialogs, which represent a SIP
relationship between a pair of user agents. Because dialogs are
between pairs of user agents, SIP's usage for two-party
communications (such as a phone call), is obvious. Communications
sessions with multiple participants, generally known as conferencing,
are more complicated. This document defines a framework for how such
conferencing can occur. This framework describes the overall
architecture, terminology, and protocol components needed for multi-
party conferencing.

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Hi Dean,

Thanks a lot for the comments. I'll have some answers inline, and will edit the draft to also reflect your suggestions.

 > -----Original Message-----
 > From: ext Dean Willis [mailto:dean.willis@softarmor.com]
 > Sent: 25 April, 2003 04:56
 > To: sipping@ietf.org; Niemi Aki (NMP/Helsinki)
 > Cc: jdrosen@dynamcisoft.com; David Oran
 > Subject: Commentary on draft-niemi-sipping-event-throttle-reqs
 > 
 > 
 > 
 > At the last meeting, we agreed (well, actually Rohan 
 > volunteered us) that we
 > would look at the event throttling requirements draft, and 
 > that at least a
 > couple of us would post discussions before we considered 
 > taking it on as a
 > WG effort. So here's my try at it.
 > 
 > The given draft is:
 > 
 > http://www.ietf.org/internet-drafts/draft-niemi-sipping-event
 > -throttle-reqs-
 > 01.txt
 > 
 > 
 > Overall, I think that developing an event throttling 
 > mechanism is a good
 > idea, and that this draft is a reasonable start on the requirements.

Good.
 
 > General comment: The draft seems to confuse the maximum 
 > notification rate
 > "desired" or "allowed" by the subscriber with the rate 
 > "required" by the
 > notifier. I'll come back to this at several points in the 
 > discussion. There
 > also seems to be a bit of confusion on the difference 
 > between average rates
 > and minimum interval -- they're not exactly inverses of each 
 > other. English
 > is such a fun language -- too bad it doesn't work well for 
 > specifications.

Good point. That certainly wasn't the intention. Simply without the require/allowed/desired semantics, a rate is the inverse of time and that was supposed to be the extent of it, and even that proved confusing... ;)

I'll try to clarify the language in the next version.
 
 > Section 3: Event Throttle Model
 > 
 > This section raises two throttling models, which it 
 > identifies as "strict"
 > and "leaky bucket" quarantines. 
 > 
 > The usage of the "strict" model should be clarified as 
 > establishing the
 > minimum time period allowed between two notifications.

I'll add text to that. 

 > The 
 > draft says that
 > the choice of which notification is sent is implementation dependent,
 > although it seems to assume a "newest" notification, where "newer"
 > notifications replace "older" ones if the minimum 
 > notification interval time
 > has not elapsed. I suspect that this replacement model has a 
 > lot of overlap
 > with filtering. Should we explicitly include a requirement that the
 > replacement policy be describable in the subscription process?

The draft actually tries not to make that assumption (but fails obviously ;). It lists as examples the replacement you also mention, as well as a merging of notifications. 

But I don't quite understand how this would overlap with filtering, as it is internal to the notifier, and depends on the type and contents of a notification, e.g., is it stateless or stateful? I think adding a requirement that the subscriber needs to be able to dictate the policy regarding the treatment of notifications in the queue would actually create that overlap.

In this case I would rather view filtering orthogonal to throttling.
 
 > This section also refers to what it calls a "leaky bucket 
 > style algorithm",
 > which is probably a misnomer. Traditionally (at least in my 
 > pointy head) a
 > "leaky bucket" refers to a FIFO queue with a variable 
 > arrival rate and a
 > fixed departure rate. There might be input "bursts" at a 
 > rate higher than
 > the output rate, resulting in a lengthening of the queue, 
 > but the output
 > rate is held constant as long as the queue length is 
 > non-zero. However, the
 > policy described in the draft is one of rate-averaging, 
 > which is something
 > like a leaky-bucket that kicks in only if the sliding-window 
 > average rate
 > exceeds the prescribed maximum. This could use some cleanup. 

Agreed. My treatment of queuing theory was obviously a bit rusty. ;)
 
 > Additionally, one might envision a rate-averaging queue that 
 > enforces its
 > operation by discarding or combining notifications instead 
 > of delaying them.

This is what the leaky-bucket was supposed to describe. Will change the draft to reflect this.
 
 > So basically, there are two queue models proposed: 
 > 
 > 1) minimum fixed interval or period between notifications, where the
 > interval can be described as a time value like 5 seconds. 
 > This can also be
 > described as maximum fixed rate, or notifications per unit 
 > of time, like 0.2
 > notifications per second or 1 notification per 5 seconds 
 > (oddly enough, this
 > is the reciprocal of the minimum interval)
 > 
 > 2) minimum average interval or period between notifications, 
 > which is better
 > described as the maximum average rate of notifications and 
 > best described in
 > notifications per unit of time, averaged over some time window.

Agreed.
 
 > And two reasonable queue-servicing policies:
 > 
 > 1) Delay notifications in a FIFO manner to conform to policy.
 > 
 > 2) Discard out-of-profile notifications or combine them with other
 > notifications to result in a notification stream that 
 > conforms to policy.

Agreed.
 
 > Note that policy 1 can result in increasing queue depth if 
 > the arrival rate
 > persists in exceeding the departure rate. Consequently, 
 > practical systems
 > using 1 are likely to revert to 2 at some point. Efficiency 
 > may also be
 > improvable by using a quenching mechanism to reduce the arrival rate.

Right, and also a variable rate policy would be likely to employ both 1 and 2 to achieve the rate averaging queue.
  
 > Section 4, Example Use Case:
 > 
 > This doesn't really describe a use case, but just presents 
 > us with the
 > mathematical notation of the reciprocal relationship between 
 > frequency and
 > period. Useful background, but a real use case (with actors, 
 > etc.) would be
 > nice here too. 
 > 
 > Perhaps something like:
 > 
 > Case 1: Alice subscribes to the status of Bob. Bob's status 
 > is served by
 > server S. Alice tells S (subscribes with a notification 
 > throttling policy)
 > that she wishes to receive notifications no more frequently 
 > than once every
 > five seconds. Bob publishes a status of Red to the server S, 
 > which notifies
 > Alice. 
 > 
 > Sub case a: After a delay of three seconds, Bob publishes a 
 > status of Green
 > to S. S, which implements a policy of discarding out-of-profile
 > notifications, stores the status of Green but does not 
 > notify Alice. After
 > three more seconds, Bob publishes a status of Blue to S. 
 > Since six seconds
 > have elapsed since the last notification to Alice, S 
 > notifies Alice that
 > Bob's status is now Blue. Note that in this sub-case, Alice is never
 > notified that Bob's status was Green, but the notifications 
 > that she does
 > receive are timely.
 > 
 > Sub case b: After a delay of three seconds, Bob publishes a 
 > status of Green
 > to S. S, which implements of policy of delaying 
 > out-of-profile notifications
 > so that they become compliant, stores the status of Bob. 
 > After two more
 > seconds, S notifies Alice that Bob's status is now Green. 
 > One second later,
 > Bob publishes a status of Blue. Again, S stores the new 
 > state for Bob, but
 > does not immediately notify Alice. Four seconds later, S 
 > notifies Alice that
 > Bob's status is now Blue. Note that in this sub-case, Alice 
 > is notified of
 > all statuses published by Bob, but often has a different 
 > understanding of
 > Bob's status than does Bob, because the notifications from S 
 > to Alice are
 > delayed.

I will add such a use-case to the draft.
  
 > REQ3: 
 > 
 > Says that "The notifier must be able to indicate that it 
 > does not support
 > the use of a throttle mechanism in the subscription. This 
 > would be better
 > worded as ". . . for a specific subscription", otherwise 
 > we'll have people
 > thinking "in the subscription" means "in the SUBSCRIBE", 
 > when we all know it
 > would have to be "in the response to the SUBSCRIBE".

Fixed.
 
 > REQ6: 
 > 
 > Ends in "For example, local policy could dictate an even 
 > lower rate of
 > notification than what the subscriber requires." Refer back 
 > to my General
 > Comment earlier in this missive. It seems to me that this 
 > would be better
 > worded as " . . . what the subscriber allows". I think we're having a
 > tendency to get confused about what the rate limit is asking for. The
 > subscriber is saying "I want to receive notifications no 
 > more frequently
 > than once per X seconds", not "You MUST notify me EVERY X seconds."

Good point. The intention was definitely not "MUST notify every X seconds". Will fix the wording.

 > REQ7:
 > 
 > Says that security mechanisms "SHOULD be applied to 
 > subscriptions . . . ".
 > This is trying to define operational guidelines at protocol 
 > requirements
 > time. Rather, we should word this sort of thing to require a 
 > capability to
 > be met by the mechanism to be developed. Perhaps "A throttle 
 > mechanism MUST
 > allow for the application of authentication and integrity protection
 > mechanisms to subscriptions invoking that mechanism."

Sounds all right to me.
 
 > 
 > Section 6: Security Requirements
 > 
 > Issue 1 points out that it might be possible for a 
 > subscriber to set a
 > maximum notification interval longer than the subscription duration,
 > consequently resulting in only getting one notification per 
 > subscription.
 > Should we add a new requirement on the order of "A notifier 
 > receiving a
 > subscription request including throttling policy MUST be 
 > able to response
 > with something having the semantic of 'That's a really 
 > freaking stupid
 > idea'.? Or should we add a requirement more like "A 
 > throttling mechanism
 > MUST include discussion of the situation resulting from a minimum
 > inter-notification period which exceeds the subscription duration and
 > include mechanisms for avoiding this situation and/or guidance to
 > implementers on the dangers of getting exactly what they asked for"?
 > 
 > Issue 2 raises the possibility that a subscriber might set a 
 > very short
 > minimum inter-notification period (or very high maximum 
 > notification rate),
 > thereby causing the notifier to exhaust itself sending at 
 > that frequency.
 > This is, well, bogus. As I pointed out above, the throttling 
 > policy does not
 > tell the notifier the rate at which notifications should be 
 > sent. Rather, it
 > tells the server the rate that notifications must not exceed. This is
 > something like a traffic sign. If the sign on the street 
 > says "Speed Limit:
 > 30 kph", this does not force all cards to speed up to 30 kph. In most
 > situations (barring a few large motorways that have a 
 > SEPARATE policy for
 > minimum speeds) driving at 28 kph, or even 20 kph, would be 
 > just fine.
 > 
 > Conclusion: Note that there is considerable potential for 
 > confusion between
 > frequency (or rate) and period (or interval) in this 
 > conversation. We should
 > be careful to say explicitly what we mean. Is it feasible to 
 > use just one
 > term in all discussions? 

Yes. I would propose that the model only talks about queues at the notifier, and minimum intervals set by a subscriber.

 > I have the feeling that SIP is not 
 > likely to be
 > well-suited to exceptionally frequent notifications. Is it 
 > reasonable to
 > state a minimum period in seconds? This lets us use a 
 > signed-16 range to
 > express inter-notification periods ranging from 1 second to 
 > a little over 9
 > hours. Do we need higher resolution than that? There's 
 > probably a resolution
 > requirement that needs to be stated in this draft.

This is probably reasonable, although a little bit into the implementation space. At some point I entertained the thought (I think Hisham suggested it), but didn't find a proper wording for such a requirement. But I'll try to come up with something proper, unless you have some text to suggest?

Cheers,
Aki
 
 > --
 > Dean
 > 
 > 
 > 
 > 
 > 
 > 
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Subject: RE: [Sipping] conf: comments on draft-rosenberg-sipping-conferenc
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>> Something is not clear to me : is CPCP really mandatory to perform
>> something as "simple" as calling many recipients via a focus (adhoc
>> conference).

> In an ad-hoc conference, the focus resides with one of the endpoints. 
> Therefore, an internal interface can be used to instruct the focus to 
> call out to additional users. So, in the case of ad-hoc, something as 
> "simple" as mass invitation does not require CPCP.

>> With a centralized conference server, a client can use a series of REFER 
>> requests to bring new users into the conference. This is described in 
>> the document as a SIP mecahnism for adding users. However, for a large 
>> number of invitations, this is really inefficient, and thus a real 
>> control protocol like CPCP is needed.

Is a CPCP body inside a SIP INVITE acceptable ? 
(I would like to avoid an additional round-trip delay for a CPCP request before the SIP request)

Then we would also have an intermediate solution for those who want that "CPCP is SIP" : CPCP would be "inside" SIP (like SDP)

>> In my opinion, it should be as simple as sending an
>> e-mail to multiple recipients (Of course, having everybody in the
>> "To" header is not possible in SIP, but having another header for
>> giving the list of recipients whould be very nice).

> Its not as simple as that. How would you report the results of that back 
> to the sender?

For me, the conference package event with SUBSCRIBE/NOTIFY would perform the job



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: End-to-end Security for Firewall/NAT Traversal within 
                          the Session Initiation Protocol (SIP)
	Author(s)	: K. Umschaden et al.
	Filename	: draft-umschaden-smime-midcom-sip-proxy-00.txt
	Pages		: 37
	Date		: 2003-5-6
	
This document describes an extension for the Session Initiation 
Protocol (SIP), which enables end-to-end security of the Session 
Description Protocol (SDP) together with firewall/Network Address 
Translation (NAT) traversal. This solution relies on Secure 
Multipurpose Internet Mail Extension (S/MIME) and the middlebox 
communications (MIDCOM) protocol. The user authorises a proxy server 
to encrypt the session description on behalf of the user. The proxy 
determines the capabilities of the receiving party and encrypts the 
SDP for a SIP proxy server in the receiving domain. Using MIDCOM, 
each proxy can dynamically control its firewall to open pinholes or 
request NAT bindings for the media flows. As long as each end-user 
may contact its trustworthy SIP proxy via a secure connection and 
authorise this proxy to encrypt the signalling data, the session 
information is secured end-to-end.

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From mailnull@www1.ietf.org  Wed May  7 18:15:05 2003
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Thread-Topic: Request for Technical Comments on ITU H.350 sipIdentity
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To Members of IETF SIPPING:

The Video Development Initiative (ViDe www.vide.net) and Internet2 Video
Middleware Working Group (VidMid http://middleware.internet2.edu/video/)
would like to call your attention to a set of ITU-T documents developed
by ITU-T Study Group 16 and known as "Directory Services Architecture
for Multimedia Conferencing"  (H.350).  
 
This ITU Recommendation describes a directory services architecture for
multimedia conferencing using LDAP. Directory services can provide
automated configuration of endpoints, association of persons with
endpoints, searchable white pages, clickable, dialing, and user
authentication based on authoritative data sources. This document
describes a standardized LDAP schema to represent endpoints on the
network and associate those endpoints with users. It discusses design
and implementation considerations for the inter-relation of video and
voice-specific directories, enterprise directories, call servers and
endpoints. 

The recommendation includes a base document (H.350) and
protocol-specific components include H.350.1 (H.323); H.350.2 (h.235);
H.350.3 (H.320); H.350.4 (SIP); and H.350.5 (generic).   Current drafts
of these documents are available at
http://middleware.internet2.edu/video/docs/H.350/  ; these
recommendations are on track for ratification at the next meeting of
Study Group 16 to be held later this month.

We would very much appreciate your individual or group technical
comments, particularly regarding H.350.4 (SIP) - we have been interested
in developing a standardized directory interface for multimedia that
contains protocol-specific information but which does not replace or
revise the protocol.  

We are aware of at least one error in H.350.4 - sipURI is listed as case
insensitive, which is incorrect.

Please send your comments to my attention, and I will see that they are
forwarded to Internet2 VidMid and also to the editor of the H.350
documents.


Sincerely,  


Jill Gemmill
jgemmill@uab.edu

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Subject: [Sipping] Re: [Sip-implementors] dtmf digits using INFO ... SIP authors ... Can you answer please ?
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Hi All,

I hate to say that but SIP community doesn't agree
or think that there are lots of use and requirement for carrying
DTMF Out-of-Band.

Every month, I can see that someone asks the same question again and again
that how do we send DTMF via INFO and Out-Of-Band .

3pcc needs it, lots of service creation environment needs it and
doesn't it looks promising to have a way in SIP to do OOB
DTMF just like h245-alphanumeric in H.323.

If we think of applications,  then there are lots of them that I can
count which doesn't care about the timing info carried by RFC2833.
And if we really care about timing, then carrying RFC2833 via
notify (Again OOB) is promising.

I really do not understand if this is a political issue to not promote
SIP INFO for carrying DTMF or if someone can justify it to me
that there is no use to carry DTMF OOB.

SIP authors, can you answer why SIP INFO for carrying simple
DTMF is not a standard solution ?

Thanks,
Rohan Kumar


----- Original Message -----
From: "Attila Sipos" <AttilaS@vegastream.com>
To: <rizsanyi@neobee.net>; <sip-implementors@cs.columbia.edu>
Sent: Friday, May 09, 2003 9:50 AM
Subject: RE: [Sip-implementors] dtmf digits using INFO


>
> Hello Zsolt,
>
> I still don't think there is a fully-agreed
> standard.  However, as much as I hate to say it,
> CISCO have their own proposed standard which some
> people (like us) interoperate with:
>
>
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122
> t/122t11/ftinfo.htm
>
> Regards,
>
> Attila
>
> Attila Sipos
> Software Engineer
> <http://www.vegastream.com>
> VegaStream : A World of difference for your Integrated Communications
>
> (jo napot kivįnok)
>
>
>
> > -----Original Message-----
> > From: Rizsanyi Zsolt [mailto:rizsanyi@myrealbox.com]
> > Sent: 09 May 2003 13:41
> > To: sip-implementors@cs.columbia.edu
> > Subject: [Sip-implementors] dtmf digits using INFO
> >
> >
> > Hi!
> >
> > Could somebody refer me to a current standard or draft, on
> > how to send dtmf
> > digits using the INFO method?
> >
> > Thanks
> > Zsolt Rizsanyi
> >
> > _______________________________________________
> > Sip-implementors mailing list
> > Sip-implementors@cs.columbia.edu
> > http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
> >
>
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@cs.columbia.edu
> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
>
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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User-Agent: Microsoft-Entourage/10.1.1.2418
Date: Fri, 09 May 2003 11:27:17 -0700
From: Cullen Jennings <fluffy@cisco.com>
To: SIP Developer <sip_developer@hotmail.com>,
        Attila Sipos <AttilaS@vegastream.com>, <rizsanyi@neobee.net>,
        <sip-implementors@cs.columbia.edu>
CC: Sipping <sipping@ietf.org>
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Subject: [Sipping] Re: [Sip-implementors] dtmf digits using INFO ... SIP authors ...
 Can you answer please ?
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Have a look at the App-Interaction stuff and KPML - it's not fully baked yet
but I think it will meet the needs of these sorts of applications.


On 5/9/03 11:05 AM, "SIP Developer" <sip_developer@hotmail.com> wrote:

> Hi All,
> 
> I hate to say that but SIP community doesn't agree
> or think that there are lots of use and requirement for carrying
> DTMF Out-of-Band.
> 
> Every month, I can see that someone asks the same question again and again
> that how do we send DTMF via INFO and Out-Of-Band .
> 
> 3pcc needs it, lots of service creation environment needs it and
> doesn't it looks promising to have a way in SIP to do OOB
> DTMF just like h245-alphanumeric in H.323.
> 
> If we think of applications,  then there are lots of them that I can
> count which doesn't care about the timing info carried by RFC2833.
> And if we really care about timing, then carrying RFC2833 via
> notify (Again OOB) is promising.
> 
> I really do not understand if this is a political issue to not promote
> SIP INFO for carrying DTMF or if someone can justify it to me
> that there is no use to carry DTMF OOB.
> 
> SIP authors, can you answer why SIP INFO for carrying simple
> DTMF is not a standard solution ?
> 
> Thanks,
> Rohan Kumar
> 
> 
> ----- Original Message -----
> From: "Attila Sipos" <AttilaS@vegastream.com>
> To: <rizsanyi@neobee.net>; <sip-implementors@cs.columbia.edu>
> Sent: Friday, May 09, 2003 9:50 AM
> Subject: RE: [Sip-implementors] dtmf digits using INFO
> 
> 
>> 
>> Hello Zsolt,
>> 
>> I still don't think there is a fully-agreed
>> standard.  However, as much as I hate to say it,
>> CISCO have their own proposed standard which some
>> people (like us) interoperate with:
>> 
>> 
> http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122
>> t/122t11/ftinfo.htm
>> 
>> Regards,
>> 
>> Attila
>> 
>> Attila Sipos
>> Software Engineer
>> <http://www.vegastream.com>
>> VegaStream : A World of difference for your Integrated Communications
>> 
>> (jo napot kivįnok)
>> 
>> 
>> 
>>> -----Original Message-----
>>> From: Rizsanyi Zsolt [mailto:rizsanyi@myrealbox.com]
>>> Sent: 09 May 2003 13:41
>>> To: sip-implementors@cs.columbia.edu
>>> Subject: [Sip-implementors] dtmf digits using INFO
>>> 
>>> 
>>> Hi!
>>> 
>>> Could somebody refer me to a current standard or draft, on
>>> how to send dtmf
>>> digits using the INFO method?
>>> 
>>> Thanks
>>> Zsolt Rizsanyi
>>> 
>>> _______________________________________________
>>> Sip-implementors mailing list
>>> Sip-implementors@cs.columbia.edu
>>> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
>>> 
>> 
>> _______________________________________________
>> Sip-implementors mailing list
>> Sip-implementors@cs.columbia.edu
>> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
>> 
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@cs.columbia.edu
> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
> 

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "SIP Developer" <sip_developer@hotmail.com>
To: "Cullen Jennings" <fluffy@cisco.com>,
        "Attila Sipos" <AttilaS@vegastream.com>, <rizsanyi@neobee.net>,
        <sip-implementors@cs.columbia.edu>
Cc: "Sipping" <sipping@ietf.org>
References: <BAE14115.77E2%fluffy@cisco.com>
Subject: Re: [Sipping] Re: [Sip-implementors] dtmf digits using INFO ... SIP authors ... Can you answer please ?
Date: Fri, 9 May 2003 14:50:40 -0400
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Cullen,

There is NO POINT in baking it FOREVER for something that will cover
everything.

Imagine the burden on developers since they have to develop
SIP INFO, RFC2833 via NOTIFY, KPML and may be few more future proposals
in the absence of SIP standards for something very basic (OOB DTMF) .

Thanks,
Rohan Kumar



----- Original Message -----
From: "Cullen Jennings" <fluffy@cisco.com>
To: "SIP Developer" <sip_developer@hotmail.com>; "Attila Sipos"
<AttilaS@vegastream.com>; <rizsanyi@neobee.net>;
<sip-implementors@cs.columbia.edu>
Cc: "Sipping" <sipping@ietf.org>
Sent: Friday, May 09, 2003 2:27 PM
Subject: [Sipping] Re: [Sip-implementors] dtmf digits using INFO ... SIP
authors ... Can you answer please ?


>
> Have a look at the App-Interaction stuff and KPML - it's not fully baked
yet
> but I think it will meet the needs of these sorts of applications.
>
>
> On 5/9/03 11:05 AM, "SIP Developer" <sip_developer@hotmail.com> wrote:
>
> > Hi All,
> >
> > I hate to say that but SIP community doesn't agree
> > or think that there are lots of use and requirement for carrying
> > DTMF Out-of-Band.
> >
> > Every month, I can see that someone asks the same question again and
again
> > that how do we send DTMF via INFO and Out-Of-Band .
> >
> > 3pcc needs it, lots of service creation environment needs it and
> > doesn't it looks promising to have a way in SIP to do OOB
> > DTMF just like h245-alphanumeric in H.323.
> >
> > If we think of applications,  then there are lots of them that I can
> > count which doesn't care about the timing info carried by RFC2833.
> > And if we really care about timing, then carrying RFC2833 via
> > notify (Again OOB) is promising.
> >
> > I really do not understand if this is a political issue to not promote
> > SIP INFO for carrying DTMF or if someone can justify it to me
> > that there is no use to carry DTMF OOB.
> >
> > SIP authors, can you answer why SIP INFO for carrying simple
> > DTMF is not a standard solution ?
> >
> > Thanks,
> > Rohan Kumar
> >
> >
> > ----- Original Message -----
> > From: "Attila Sipos" <AttilaS@vegastream.com>
> > To: <rizsanyi@neobee.net>; <sip-implementors@cs.columbia.edu>
> > Sent: Friday, May 09, 2003 9:50 AM
> > Subject: RE: [Sip-implementors] dtmf digits using INFO
> >
> >
> >>
> >> Hello Zsolt,
> >>
> >> I still don't think there is a fully-agreed
> >> standard.  However, as much as I hate to say it,
> >> CISCO have their own proposed standard which some
> >> people (like us) interoperate with:
> >>
> >>
> >
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122
> >> t/122t11/ftinfo.htm
> >>
> >> Regards,
> >>
> >> Attila
> >>
> >> Attila Sipos
> >> Software Engineer
> >> <http://www.vegastream.com>
> >> VegaStream : A World of difference for your Integrated Communications
> >>
> >> (jo napot kivįnok)
> >>
> >>
> >>
> >>> -----Original Message-----
> >>> From: Rizsanyi Zsolt [mailto:rizsanyi@myrealbox.com]
> >>> Sent: 09 May 2003 13:41
> >>> To: sip-implementors@cs.columbia.edu
> >>> Subject: [Sip-implementors] dtmf digits using INFO
> >>>
> >>>
> >>> Hi!
> >>>
> >>> Could somebody refer me to a current standard or draft, on
> >>> how to send dtmf
> >>> digits using the INFO method?
> >>>
> >>> Thanks
> >>> Zsolt Rizsanyi
> >>>
> >>> _______________________________________________
> >>> Sip-implementors mailing list
> >>> Sip-implementors@cs.columbia.edu
> >>> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
> >>>
> >>
> >> _______________________________________________
> >> Sip-implementors mailing list
> >> Sip-implementors@cs.columbia.edu
> >> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
> >>
> > _______________________________________________
> > Sip-implementors mailing list
> > Sip-implementors@cs.columbia.edu
> > http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
> >
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: Re: [Sipping] Re: [Sip-implementors] dtmf digits using INFO ... SIP authors ... Can you answer please ?
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To: "SIP Developer" <sip_developer@hotmail.com>
From: Rohan Mahy <rohan@cisco.com>
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Hi Rohan,

The main reason to not carry DTMF in SIP INFOs is that there is a very  
good chance that you will get double-digits or otherwise corrupted  
digits because the signaling is not time-synchronized with the audio  
media.  This issue is discussed in an expired draft which I wrote a  
long time ago, which I abandoned in favor of the work generated by the  
app-interaction design team. At this point my draft is useful only in  
that it explains how double digits tend to occur.

my expired draft:

http://www.softarmor.com/sipping/drafts/morgue/draft-mahy-sipping- 
signaled-digits-01.html

the app-interaction design team drafts:

http://www.softarmor.com/sipping/drafts/draft-burger-sipping-kpml-01.txt
http://www.softarmor.com/sipping/drafts/draft-culpepper-sipping-app- 
interact-reqs-03.txt
http://www.softarmor.com/sipping/drafts/draft-jennings-sip-app-info- 
00.txt
http://www.softarmor.com/sipping/drafts/draft-rosenberg-sipping-app- 
interaction-framework-00.txt


thanks,
-rohan (Mahy)


On Friday, May 9, 2003, at 11:05 AM, SIP Developer wrote:

> Hi All,
>
> I hate to say that but SIP community doesn't agree
> or think that there are lots of use and requirement for carrying
> DTMF Out-of-Band.
>
> Every month, I can see that someone asks the same question again and  
> again
> that how do we send DTMF via INFO and Out-Of-Band .
>
> 3pcc needs it, lots of service creation environment needs it and
> doesn't it looks promising to have a way in SIP to do OOB
> DTMF just like h245-alphanumeric in H.323.
>
> If we think of applications,  then there are lots of them that I can
> count which doesn't care about the timing info carried by RFC2833.
> And if we really care about timing, then carrying RFC2833 via
> notify (Again OOB) is promising.
>
> I really do not understand if this is a political issue to not promote
> SIP INFO for carrying DTMF or if someone can justify it to me
> that there is no use to carry DTMF OOB.
>
> SIP authors, can you answer why SIP INFO for carrying simple
> DTMF is not a standard solution ?
>
> Thanks,
> Rohan Kumar
>
>
> ----- Original Message -----
> From: "Attila Sipos" <AttilaS@vegastream.com>
> To: <rizsanyi@neobee.net>; <sip-implementors@cs.columbia.edu>
> Sent: Friday, May 09, 2003 9:50 AM
> Subject: RE: [Sip-implementors] dtmf digits using INFO
>
>
>>
>> Hello Zsolt,
>>
>> I still don't think there is a fully-agreed
>> standard.  However, as much as I hate to say it,
>> CISCO have their own proposed standard which some
>> people (like us) interoperate with:
>>
>>
> http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/ 
> 122newft/122
>> t/122t11/ftinfo.htm
>>
>> Regards,
>>
>> Attila
>>
>> Attila Sipos
>> Software Engineer
>> <http://www.vegastream.com>
>> VegaStream : A World of difference for your Integrated Communications
>>
>> (jo napot kivįnok)
>>
>>
>>
>>> -----Original Message-----
>>> From: Rizsanyi Zsolt [mailto:rizsanyi@myrealbox.com]
>>> Sent: 09 May 2003 13:41
>>> To: sip-implementors@cs.columbia.edu
>>> Subject: [Sip-implementors] dtmf digits using INFO
>>>
>>>
>>> Hi!
>>>
>>> Could somebody refer me to a current standard or draft, on
>>> how to send dtmf
>>> digits using the INFO method?
>>>
>>> Thanks
>>> Zsolt Rizsanyi
>>>
>>> _______________________________________________
>>> Sip-implementors mailing list
>>> Sip-implementors@cs.columbia.edu
>>> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
>>>
>>
>> _______________________________________________
>> Sip-implementors mailing list
>> Sip-implementors@cs.columbia.edu
>> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
>>
> _______________________________________________
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Hello Everyone,

I'd like to begin Working Group Last Call on:

http:/www.ietf.org/internet-drafts/draft-ietf-sipping-e164-02.txt

this WGLC will end on June 13, 2003.

thanks,
-rohan

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Rohan Mahy writes:

  > I'd like to begin Working Group Last Call on:
  > 
  > http:/www.ietf.org/internet-drafts/draft-ietf-sipping-e164-02.txt
  > 
  > this WGLC will end on June 13, 2003.

i must be seeing a VERY bad dream. the draft uses the wrong service
field for sip: "sip+E2U" instead of "E2U+sip" as per
draft-ietf-enum-rfc2916bis-06.txt.

please discard the sipping draft from the archives AS SOON AS POSSIBLE
so that not a single new implementer will find and read it.

-- juha

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Rohan Mahy writes:

  > I'd like to begin Working Group Last Call on:
  > 
  > http:/www.ietf.org/internet-drafts/draft-ietf-sipping-e164-02.txt
  > 
  > this WGLC will end on June 13, 2003.

i must be seeing a VERY bad dream. the draft uses the wrong service
field for sip: "sip+E2U" instead of "E2U+sip" as per
draft-ietf-enum-rfc2916bis-06.txt.

please discard the sipping draft from the archives AS SOON AS POSSIBLE
so that not a single new implementer will find and read it.

-- juha
_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

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No, you're not suffering any bad dream - just an attack of rhetorical
exaggeration and lax reading. The authors of sipping-e164 are aware of the
rfc2916bis Internet-Draft, and have suitably referenced it in Section 7,
which is entitled "Updates for rfc2916bis". It mentions the new enumservice
format that concerns you so greatly below, and recommends that SIP
applications accept either format for purposes of backwards compatibility -
quite generous considering that rfc2916bis is just a I-D, not a standard. In
the light of all this, it is hard for me to see a justification for
banishing sipping-e164 from the I-D archives.

But granted - the last time that sipping-e164 was updated (October 2002),
rfc2916bis was a lot less mature than it is now. Given that a last call for
rfc2916bis is also now in progress, it might make sense to update
sipping-e164 to reference (normatively) the new rfc2916bis notation
throughout, assuming that they will both go before the IESG at about the
same time. A fair last call comment; we'll remedy this before sipping-e164
goes to the IESG.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Juha Heinanen [mailto:jh@lohi.eng.song.fi]
> Sent: Tuesday, May 13, 2003 10:10 PM
> To: Rohan Mahy
> Cc: 'sipping@ietf.org'; jon.peterson@neustar.biz;
> bcampbell@dynamicsoft.com; 'Dean Willis'; 
> gonzalo.Camarillo@ericsson.com
> Subject: [Sipping] WGLC for draft-ietf-sipping-e164-02.txt
> 
> 
> Rohan Mahy writes:
> 
> > I'd like to begin Working Group Last Call on:
> > 
> > http:/www.ietf.org/internet-drafts/draft-ietf-sipping-e164-02.txt
> > 
> > this WGLC will end on June 13, 2003.
> 
> i must be seeing a VERY bad dream. the draft uses the wrong service
> field for sip: "sip+E2U" instead of "E2U+sip" as per
> draft-ietf-enum-rfc2916bis-06.txt.
> 
> please discard the sipping draft from the archives AS SOON AS POSSIBLE
> so that not a single new implementer will find and read it.
> 
> -- juha
> 

14-May-2003 02:36:36 -0500,5113;000000000001-00000b05
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Juha Heinanen'" <jh@lohi.eng.song.fi>, Rohan Mahy <rohan@cisco.com>
Cc: "'sipping@ietf.org'" <sipping@ietf.org>, bcampbell@dynamicsoft.com,
      "'Dean Willis'" <dean.willis@softarmor.com>, gonzalo.Camarillo@ericsson.com
Subject: RE: [Sipping] WGLC for draft-ietf-sipping-e164-02.txt
Date: Wed, 14 May 2003 03:27:13 -0400
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No, you're not suffering any bad dream - just an attack of rhetorical
exaggeration and lax reading. The authors of sipping-e164 are aware of the
rfc2916bis Internet-Draft, and have suitably referenced it in Section 7,
which is entitled "Updates for rfc2916bis". It mentions the new enumservice
format that concerns you so greatly below, and recommends that SIP
applications accept either format for purposes of backwards compatibility -
quite generous considering that rfc2916bis is just a I-D, not a standard. In
the light of all this, it is hard for me to see a justification for
banishing sipping-e164 from the I-D archives.

But granted - the last time that sipping-e164 was updated (October 2002),
rfc2916bis was a lot less mature than it is now. Given that a last call for
rfc2916bis is also now in progress, it might make sense to update
sipping-e164 to reference (normatively) the new rfc2916bis notation
throughout, assuming that they will both go before the IESG at about the
same time. A fair last call comment; we'll remedy this before sipping-e164
goes to the IESG.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Juha Heinanen [mailto:jh@lohi.eng.song.fi]
> Sent: Tuesday, May 13, 2003 10:10 PM
> To: Rohan Mahy
> Cc: 'sipping@ietf.org'; jon.peterson@neustar.biz;
> bcampbell@dynamicsoft.com; 'Dean Willis'; 
> gonzalo.Camarillo@ericsson.com
> Subject: [Sipping] WGLC for draft-ietf-sipping-e164-02.txt
> 
> 
> Rohan Mahy writes:
> 
> > I'd like to begin Working Group Last Call on:
> > 
> > http:/www.ietf.org/internet-drafts/draft-ietf-sipping-e164-02.txt
> > 
> > this WGLC will end on June 13, 2003.
> 
> i must be seeing a VERY bad dream. the draft uses the wrong service
> field for sip: "sip+E2U" instead of "E2U+sip" as per
> draft-ietf-enum-rfc2916bis-06.txt.
> 
> please discard the sipping draft from the archives AS SOON AS POSSIBLE
> so that not a single new implementer will find and read it.
> 
> -- juha
> 
_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

14-May-2003 03:56:44 -0500,3487;000000000001-00000b06
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From: "Gonzalo Camarillo Gonzalez (LMF)"
                  <Gonzalo.Camarillo@lmf.ericsson.se>
To: "'sipping@ietf.org'" <sipping@ietf.org>
Cc: "Jon Peterson (E-mail)" <jon.peterson@neustar.biz>,
      "Rohan Mahy (E-mail)" <rohan@cisco.com>,
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Subject: [Sipping] Last Call for early media use cases
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Folks,

in the last IETF meeting, we requested early media use cases that show the need of a "media coming" flag. So far, we have not received anything.

We intend to release a new version of the early media draft in a week or so. So, that is the time you have to send me your use cases.

Thanks,

Gonzalo
_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

14-May-2003 13:35:23 -0500,6494;000000000001-00000b07
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From: "Mark Watson" <mwatson@nortelnetworks.com>
To: "'Gonzalo Camarillo Gonzalez (LMF)'"
                  <Gonzalo.Camarillo@lmf.ericsson.se>,
      "'sipping@ietf.org'"
                  <sipping@ietf.org>
Cc: "Jon Peterson (E-mail)" <jon.peterson@neustar.biz>,
      "Rohan Mahy (E-mail)" <rohan@cisco.com>,
      "Dean Willis (E-mail)"
                  <dean.willis@softarmor.com>
Subject: RE: [Sipping] Last Call for early media use cases
Date: Wed, 14 May 2003 19:32:08 +0100
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------_=3D_NextPart_001_01C31A47.1F6320FA
Gonzalo,

I think you have received plenty of emails on this topic. What is clear =
is
that there is no consensus either way that such an indication is or is =
not
required.

Anyway, for my part I do not have any further information I can provide =
you
that I think has any prospect of convincing you. I still believe the =
absence
of some kind of additional indication will in practice lead to a
less-than-ideal user experience, but I quite accept there is no =
consensus
right now.

Regards,

Mark

> -----Original Message-----
> From: Gonzalo Camarillo Gonzalez (LMF)
> [mailto:Gonzalo.Camarillo@lmf.ericsson.se]
> Sent: 14 May 2003 00:13
> To: 'sipping@ietf.org'
> Cc: Jon Peterson (E-mail); Rohan Mahy (E-mail); Dean Willis (E-mail)
> Subject: [Sipping] Last Call for early media use cases
>=20
>=20
> Folks,
>=20
> in the last IETF meeting, we requested early media use cases=20
> that show the need of a "media coming" flag. So far, we have=20
> not received anything.
>=20
> We intend to release a new version of the early media draft=20
> in a week or so. So, that is the time you have to send me=20
> your use cases.
>=20
> Thanks,
>=20
> Gonzalo
> _______________________________________________
> Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20

------_=3D_NextPart_001_01C31A47.1F6320FA
[Enclosed text/html ]
------_=3D_NextPart_001_01C31A47.1F6320FA--

14-May-2003 13:44:38 -0500,8197;000000000001-00000b08
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To: "'Gonzalo Camarillo Gonzalez (LMF)'"
                  <Gonzalo.Camarillo@lmf.ericsson.se>,
      "'sipping@ietf.org'"
                  <sipping@ietf.org>
Cc: "Jon Peterson (E-mail)" <jon.peterson@neustar.biz>,
      "Rohan Mahy (E-mail)" <rohan@cisco.com>,
      "Dean Willis (E-mail)"
                  <dean.willis@softarmor.com>
Subject: RE: [Sipping] Last Call for early media use cases
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------_=3D_NextPart_001_01C31A47.1F6320FA
Gonzalo,

I think you have received plenty of emails on this topic. What is clear =
is
that there is no consensus either way that such an indication is or is =
not
required.

Anyway, for my part I do not have any further information I can provide =
you
that I think has any prospect of convincing you. I still believe the =
absence
of some kind of additional indication will in practice lead to a
less-than-ideal user experience, but I quite accept there is no =
consensus
right now.

Regards,

Mark

> -----Original Message-----
> From: Gonzalo Camarillo Gonzalez (LMF)
> [mailto:Gonzalo.Camarillo@lmf.ericsson.se]
> Sent: 14 May 2003 00:13
> To: 'sipping@ietf.org'
> Cc: Jon Peterson (E-mail); Rohan Mahy (E-mail); Dean Willis (E-mail)
> Subject: [Sipping] Last Call for early media use cases
>=20
>=20
> Folks,
>=20
> in the last IETF meeting, we requested early media use cases=20
> that show the need of a "media coming" flag. So far, we have=20
> not received anything.
>=20
> We intend to release a new version of the early media draft=20
> in a week or so. So, that is the time you have to send me=20
> your use cases.
>=20
> Thanks,
>=20
> Gonzalo
> _______________________________________________
> Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20

------_=3D_NextPart_001_01C31A47.1F6320FA
[Enclosed text/html ]
------_=3D_NextPart_001_01C31A47.1F6320FA--
_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

14-May-2003 16:51:39 -0500,5088;000000000001-00000b09
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To: "Peterson, Jon" <jon.peterson@neustar.biz>,
      "'Juha Heinanen'"
                  <jh@lohi.eng.song.fi>, Rohan Mahy <rohan@cisco.com>
Cc: "'sipping@ietf.org'" <sipping@ietf.org>, bcampbell@dynamicsoft.com,
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Date: Wed, 14 May 2003 22:51:03 +0100
Subject: RE: [Sipping] WGLC for draft-ietf-sipping-e164-02.txt
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At 3:27 am -0400 14/5/03, Peterson, Jon wrote:
>No, you're not suffering any bad dream - just an attack of rhetorical
>exaggeration and lax reading. The authors of sipping-e164 are aware of =
the
>rfc2916bis Internet-Draft, and have suitably referenced it in Section =
7,
>which is entitled "Updates for rfc2916bis". It mentions the new =
enumservice
>format that concerns you so greatly below, and recommends that SIP
>applications accept either format for purposes of backwards =
compatibility -
>quite generous considering that rfc2916bis is just a I-D, not a =
standard. In
>the light of all this, it is hard for me to see a justification for
>banishing sipping-e164 from the I-D archives.
>
>But granted - the last time that sipping-e164 was updated (October =
2002),
>rfc2916bis was a lot less mature than it is now. Given that a last =
call for
>rfc2916bis is also now in progress, it might make sense to update
>sipping-e164 to reference (normatively) the new rfc2916bis notation
>throughout, assuming that they will both go before the IESG at about =
the
>same time. A fair last call comment; we'll remedy this before =
sipping-e164
>goes to the IESG.
>
>Jon Peterson
>NeuStar, Inc.
>
>Juha Wrote:
> > i must be seeing a VERY bad dream. the draft uses the wrong =
service
>> field for sip: "sip+E2U" instead of "E2U+sip" as per
>> draft-ietf-enum-rfc2916bis-06.txt.
>>
>> please discard the sipping draft from the archives AS SOON AS =
POSSIBLE
> > so that not a single new implementer will find and read it.
> >
To which I reply:

Hi folks,
    RFC2916 is going to be obsoleted by rfc2916bis-06 (if approved)? =
Thus,
this backward compatibility in sipping-e164 would be with be with an =
obsoleted
document (not even a current I-D).

I have a lot of sympathy with Juha on this one - we have already had=20
fun with one
bunch (strictly) implementing 2916 whilst the other bunch (strictly) =
implement
2916bis - I'd prefer not continuing this. Was this a VERY bad dream?=20
- I fear not.

=3D> Do you -REALLY- -REALLY- need backward compatibility?

My vote's to just go with the 2916bis syntax throughout, and remove =
reference
to the soon to be obsoleted syntax in sipping-164?

BTW, will we see an up-issued version as -02 IS out of date and has =
some things
that will/may/might change?

all the best,
      Lawrence
--=20
-----------------------------------------------------------------------
Roke Manor Research : This information is provided "as is" and is =
not
<mailto:lwc@roke.co.uk>: intended to create any contractual or legal
<tel:+441794833666> : relationship.

--------------InterScan_NT_MIME_Boundary
Registered Office: Roke Manor Research Ltd, Siemens House, Oldbury, =
Bracknell,=20
Berkshire. RG12 8FZ

The information contained in this e-mail and any attachments is =
confidential to Roke Manor Research Ltd and must not be passed to any =
third party without permission. This communication is for information =
only and shall not create or change any contractual relationship.

--------------InterScan_NT_MIME_Boundary--

14-May-2003 17:05:05 -0500,5976;000000000001-00000b0a
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To: "Peterson, Jon" <jon.peterson@neustar.biz>,
      "'Juha Heinanen'"
                  <jh@lohi.eng.song.fi>, Rohan Mahy <rohan@cisco.com>
Cc: "'sipping@ietf.org'" <sipping@ietf.org>, bcampbell@dynamicsoft.com,
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  <0449D80A0E9B614A83FA9031B07E8D3B257A6B@stntexch2.va.neustar.com>
Date: Wed, 14 May 2003 22:51:03 +0100
Subject: RE: [Sipping] WGLC for draft-ietf-sipping-e164-02.txt
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At 3:27 am -0400 14/5/03, Peterson, Jon wrote:
>No, you're not suffering any bad dream - just an attack of rhetorical
>exaggeration and lax reading. The authors of sipping-e164 are aware of =
the
>rfc2916bis Internet-Draft, and have suitably referenced it in Section =
7,
>which is entitled "Updates for rfc2916bis". It mentions the new =
enumservice
>format that concerns you so greatly below, and recommends that SIP
>applications accept either format for purposes of backwards =
compatibility -
>quite generous considering that rfc2916bis is just a I-D, not a =
standard. In
>the light of all this, it is hard for me to see a justification for
>banishing sipping-e164 from the I-D archives.
>
>But granted - the last time that sipping-e164 was updated (October =
2002),
>rfc2916bis was a lot less mature than it is now. Given that a last =
call for
>rfc2916bis is also now in progress, it might make sense to update
>sipping-e164 to reference (normatively) the new rfc2916bis notation
>throughout, assuming that they will both go before the IESG at about =
the
>same time. A fair last call comment; we'll remedy this before =
sipping-e164
>goes to the IESG.
>
>Jon Peterson
>NeuStar, Inc.
>
>Juha Wrote:
> > i must be seeing a VERY bad dream. the draft uses the wrong =
service
>> field for sip: "sip+E2U" instead of "E2U+sip" as per
>> draft-ietf-enum-rfc2916bis-06.txt.
>>
>> please discard the sipping draft from the archives AS SOON AS =
POSSIBLE
> > so that not a single new implementer will find and read it.
> >
To which I reply:

Hi folks,
    RFC2916 is going to be obsoleted by rfc2916bis-06 (if approved)? =
Thus,
this backward compatibility in sipping-e164 would be with be with an =
obsoleted
document (not even a current I-D).

I have a lot of sympathy with Juha on this one - we have already had=20
fun with one
bunch (strictly) implementing 2916 whilst the other bunch (strictly) =
implement
2916bis - I'd prefer not continuing this. Was this a VERY bad dream?=20
- I fear not.

=3D> Do you -REALLY- -REALLY- need backward compatibility?

My vote's to just go with the 2916bis syntax throughout, and remove =
reference
to the soon to be obsoleted syntax in sipping-164?

BTW, will we see an up-issued version as -02 IS out of date and has =
some things
that will/may/might change?

all the best,
      Lawrence
--=20
-----------------------------------------------------------------------
Roke Manor Research : This information is provided "as is" and is =
not
<mailto:lwc@roke.co.uk>: intended to create any contractual or legal
<tel:+441794833666> : relationship.
_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

14-May-2003 20:57:18 -0500,8136;000000000001-00000b0b
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------=3D_NextPart_000_0043_01C31ACF.4F688C30
Hello, 

Nokia, Siemens and Ericsson have announced tu”ŠŃ”äÉŁ±½Į„¹ĮÕĶ µŃ¼µŃ±¬ĮĮ±„Ń„½ø½ČMŌŅęWGv÷&·2vFu%2ęBf÷"gWGW&R4ręWGv÷&·2āFWvē]Č\ŁHŅTÜŚYŪ[[Č]HŪŻ[Ż[[H[ÜHetailed information. 

Does someone of you know some detu„°½ÕŠŃ”AQPĶ„¹±„¹Ż„Ń M%@ü4)¼Ń”äÉ½ÕŃŅ&&G&FöābGvņV÷ĘRvēBFņFĘ²FņFR6ÖRw&÷WH]HŲ[YH[YOČ\H\HŪŪYHYČÜ\ŲŻ\ÜŚ[ŪČ]bout this topics? 

regards, 
thomas lee. 

          
        tQ”½µĢ1Q”¹„° ½¹ĶÕ±Ń¹ŃA½±åĮ„ą%¹ĶŃ„ŃÕŃŅöbFV6ęöĘöwb7VęvFņfVēGW&RF÷vW"ĀcRÓ"6×7VęrÖFöęwKŲ[Ū[KZŻKŁ[Ż[ LĶKN KŪÜXU[
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_______________________________________________
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Use sip@ietf.org for new developments of core SIP

14-May-2003 21:32:56 -0500,11696;000000000001-00000b0c
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Subject: [Sipping] Questions and comments on =
draft-rosenberg-sipping-ice-00
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------_=3D_NextPart_001_01C31A88.B82957DC
Hi Jonathan,

Better late than never I guess. I have a few questions/comments on=20
draft-rosenberg-sipping-ice-00.html.

- Section 2 (the overview) says that the UAC should attempt to derive
    IP addresses with STUN, TURN, RSIP and VPN servers and place all of=20
    these addresses in the offer SDP. Sections 3, 5.1 and 5.4 confirm =
this=20
    procedure. In the case of STUN, it means a dedicated STUN server as
    opposed to the peer-to-peer UA STUN server used at call setup. =
However,
    section 12.3 may lead the reader to believe that a dedicated STUN=20
    server) is "simply not used" and that instead the peer-to-peer UA =
STUN
    servers are sufficient. The wording should be re-done to clarify that
    what is not used is the NAT classification process (the derivation of =

    the IP address using a dedicated STUN server is indeed used). At =
least,
    I think this is what 12.3 is saying. If there was an example 9.3,=20
    similar to 9.2 but where B calls A, I assume that B would include =
both=20
    LTA2 and PDTA2 in the offer, and that PDTA2 would be marked as
    derived. Is this correct? Since the 2 examples of section 9 do NOT =
have
    a STUN-derived address for the calling party, it is even more =
confusing.

- Section 7.2 says that the derived-attribute refer to transport =
addresses
    "learned from the entity to whom the SDP is being sent (referred to
    as the peer)". This is the same issues as the previous one. I don't =
think
    this is correct. It is the transport address learned from the STUN
    query: in the initial offer, that address is NOT learned from the
    "peer" but from the dedicated STUN server. Correct? In the answer or =
in
    a second round of offer (in an UPDATE), then is is indeed learned =
from=20
    the peer. Is this correct?

- In examples 9.1 and 9.2, A only has one address, a local IPv4 =
address.=20
    The qvalue is set to 1.0 in the offer in both examples. According to=20
    the rules 5.3, shouldn't the qvalue in the offer been set to 0.8=20
    instead?

- In example 9.1, can't we skip the whole peer-to-peer STUN query since
    both A and B know that no NATs are involved from their previous STUN=20
    query to the dedicated STUN servers? I mean, when B receives A's
    offer, it sees only one address (non-derived), and it knows that
    itself has only a non-derived IP address.

- In a few places (including the abstract), it is claimed that ICE is =
less
    brittle than STUN. This may be a little misleading since ICE itselfs
    uses the STUN protocol. It should be clarified (as per my earlier =
point),=20
    that what is meant by "STUN" in this context is that traditional STUN =
only

    is more brittle than ICE.


Typos:
- Section 2, third paragraph, first sentence. Replace "realsm" with=20
    "realms".

- In section 9.2, in the paragraph right before the description of the =

    UPDATE content, replace "has no offered" with "has not offered".


----
Fran=3DE7ois AUDET, Nortel Networks
Tel: +1 408 495 3756

------_=3D_NextPart_001_01C31A88.B82957DC
[Enclosed text/html ]
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_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

15-May-2003 07:06:33 -0500,5234;000000000001-00000b0d
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Conroy, Lawrence (SMTP)'" <lwc@roke.co.uk>,
      "'Juha Heinanen'"
                  <jh@lohi.eng.song.fi>, Rohan Mahy <rohan@cisco.com>
Cc: "'sipping@ietf.org'" <sipping@ietf.org>, bcampbell@dynamicsoft.com,
      "'Dean Willis'" <dean.willis@softarmor.com>, =
gonzalo.Camarillo@ericsson.com
Subject: RE: [Sipping] WGLC for draft-ietf-sipping-e164-02.txt
Date: Thu, 15 May 2003 08:06:43 -0400
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As I indicated in my last note, there will be an updated version, yes, =
but
it will not entirely neglect the soon-to-be-superannuated RFC2916. I =
don't
think it is too much to ask that SIP applications utilizing ENUM be =
generous
in what they receive with regard to enumservices - it just doesn't =
seem
costly to be backwards compatible with the original iteration of the =
ENUM
standard.

If there are things other than the enumservice update that people would =
like
to see mended, this last call period would be a good time to let the =
authors
know.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Conroy, Lawrence (SMTP) [mailto:lwc@roke.co.uk]
> Sent: Wednesday, May 14, 2003 2:51 PM
> To: Peterson, Jon; 'Juha Heinanen'; Rohan Mahy
> Cc: 'sipping@ietf.org'; bcampbell@dynamicsoft.com; 'Dean Willis';
> gonzalo.Camarillo@ericsson.com
> Subject: RE: [Sipping] WGLC for draft-ietf-sipping-e164-02.txt
>=20
>=20
> At 3:27 am -0400 14/5/03, Peterson, Jon wrote:
> >No, you're not suffering any bad dream - just an attack of =
rhetorical
> >exaggeration and lax reading. The authors of sipping-e164=20
> are aware of the
> >rfc2916bis Internet-Draft, and have suitably referenced it=20
> in Section 7,
> >which is entitled "Updates for rfc2916bis". It mentions the=20
> new enumservice
> >format that concerns you so greatly below, and recommends that SIP
> >applications accept either format for purposes of backwards=20
> compatibility -
> >quite generous considering that rfc2916bis is just a I-D,=20
> not a standard. In
> >the light of all this, it is hard for me to see a justification for
> >banishing sipping-e164 from the I-D archives.
> >
> >But granted - the last time that sipping-e164 was updated=20
> (October 2002),
> >rfc2916bis was a lot less mature than it is now. Given that=20
> a last call for
> >rfc2916bis is also now in progress, it might make sense to update
> >sipping-e164 to reference (normatively) the new rfc2916bis notation
> >throughout, assuming that they will both go before the IESG=20
> at about the
> >same time. A fair last call comment; we'll remedy this=20
> before sipping-e164
> >goes to the IESG.
> >
> >Jon Peterson
> >NeuStar, Inc.
> >
> >Juha Wrote:
> > > i must be seeing a VERY bad dream. the draft uses the=20
> wrong service
> >> field for sip: "sip+E2U" instead of "E2U+sip" as per
> >> draft-ietf-enum-rfc2916bis-06.txt.
> >>
> >> please discard the sipping draft from the archives AS=20
> SOON AS POSSIBLE
> > > so that not a single new implementer will find and read it.
> > >
> To which I reply:
>=20
> Hi folks,
> RFC2916 is going to be obsoleted by rfc2916bis-06 (if=20
> approved)? Thus,
> this backward compatibility in sipping-e164 would be with be=20
> with an obsoleted
> document (not even a current I-D).
>=20
> I have a lot of sympathy with Juha on this one - we have already had=20
> fun with one
> bunch (strictly) implementing 2916 whilst the other bunch=20
> (strictly) implement
> 2916bis - I'd prefer not continuing this. Was this a VERY bad dream?=20
> - I fear not.
>=20
> =3D> Do you -REALLY- -REALLY- need backward compatibility?
>=20
> My vote's to just go with the 2916bis syntax throughout, and=20
> remove reference
> to the soon to be obsoleted syntax in sipping-164?
>=20
> BTW, will we see an up-issued version as -02 IS out of date=20
> and has some things
> that will/may/might change?
>=20
> all the best,
> Lawrence
> --=20
> --------------------------------------------------------------
> ---------
> Roke Manor Research : This information is provided "as is"=20
> and is not
> <mailto:lwc@roke.co.uk>: intended to create any contractual or legal
> <tel:+441794833666> : relationship.
>=20

15-May-2003 07:17:36 -0500,6901;000000000001-00000b0e
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<0449D80A0E9B614A83FA9031B07E8D3B257A78@stntexch2.va.neustar.com>
From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Conroy, Lawrence (SMTP)'" <lwc@roke.co.uk>,
      "'Juha Heinanen'"
                  <jh@lohi.eng.song.fi>, Rohan Mahy <rohan@cisco.com>
Cc: "'sipping@ietf.org'" <sipping@ietf.org>, bcampbell@dynamicsoft.com,
      "'Dean Willis'" <dean.willis@softarmor.com>, =
gonzalo.Camarillo@ericsson.com
Subject: RE: [Sipping] WGLC for draft-ietf-sipping-e164-02.txt
Date: Thu, 15 May 2003 08:06:43 -0400
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As I indicated in my last note, there will be an updated version, yes, =
but
it will not entirely neglect the soon-to-be-superannuated RFC2916. I =
don't
think it is too much to ask that SIP applications utilizing ENUM be =
generous
in what they receive with regard to enumservices - it just doesn't =
seem
costly to be backwards compatible with the original iteration of the =
ENUM
standard.

If there are things other than the enumservice update that people would =
like
to see mended, this last call period would be a good time to let the =
authors
know.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Conroy, Lawrence (SMTP) [mailto:lwc@roke.co.uk]
> Sent: Wednesday, May 14, 2003 2:51 PM
> To: Peterson, Jon; 'Juha Heinanen'; Rohan Mahy
> Cc: 'sipping@ietf.org'; bcampbell@dynamicsoft.com; 'Dean Willis';
> gonzalo.Camarillo@ericsson.com
> Subject: RE: [Sipping] WGLC for draft-ietf-sipping-e164-02.txt
>=20
>=20
> At 3:27 am -0400 14/5/03, Peterson, Jon wrote:
> >No, you're not suffering any bad dream - just an attack of =
rhetorical
> >exaggeration and lax reading. The authors of sipping-e164=20
> are aware of the
> >rfc2916bis Internet-Draft, and have suitably referenced it=20
> in Section 7,
> >which is entitled "Updates for rfc2916bis". It mentions the=20
> new enumservice
> >format that concerns you so greatly below, and recommends that SIP
> >applications accept either format for purposes of backwards=20
> compatibility -
> >quite generous considering that rfc2916bis is just a I-D,=20
> not a standard. In
> >the light of all this, it is hard for me to see a justification for
> >banishing sipping-e164 from the I-D archives.
> >
> >But granted - the last time that sipping-e164 was updated=20
> (October 2002),
> >rfc2916bis was a lot less mature than it is now. Given that=20
> a last call for
> >rfc2916bis is also now in progress, it might make sense to update
> >sipping-e164 to reference (normatively) the new rfc2916bis notation
> >throughout, assuming that they will both go before the IESG=20
> at about the
> >same time. A fair last call comment; we'll remedy this=20
> before sipping-e164
> >goes to the IESG.
> >
> >Jon Peterson
> >NeuStar, Inc.
> >
> >Juha Wrote:
> > > i must be seeing a VERY bad dream. the draft uses the=20
> wrong service
> >> field for sip: "sip+E2U" instead of "E2U+sip" as per
> >> draft-ietf-enum-rfc2916bis-06.txt.
> >>
> >> please discard the sipping draft from the archives AS=20
> SOON AS POSSIBLE
> > > so that not a single new implementer will find and read it.
> > >
> To which I reply:
>=20
> Hi folks,
> RFC2916 is going to be obsoleted by rfc2916bis-06 (if=20
> approved)? Thus,
> this backward compatibility in sipping-e164 would be with be=20
> with an obsoleted
> document (not even a current I-D).
>=20
> I have a lot of sympathy with Juha on this one - we have already had=20
> fun with one
> bunch (strictly) implementing 2916 whilst the other bunch=20
> (strictly) implement
> 2916bis - I'd prefer not continuing this. Was this a VERY bad dream?=20
> - I fear not.
>=20
> =3D> Do you -REALLY- -REALLY- need backward compatibility?
>=20
> My vote's to just go with the 2916bis syntax throughout, and=20
> remove reference
> to the soon to be obsoleted syntax in sipping-164?
>=20
> BTW, will we see an up-issued version as -02 IS out of date=20
> and has some things
> that will/may/might change?
>=20
> all the best,
> Lawrence
> --=20
> --------------------------------------------------------------
> ---------
> Roke Manor Research : This information is provided "as is"=20
> and is not
> <mailto:lwc@roke.co.uk>: intended to create any contractual or legal
> <tel:+441794833666> : relationship.
>=20
_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

15-May-2003 08:06:56 -0500,1952;000000000001-00000b0f
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Date: Thu, 15 May 2003 16:06:24 +0300
To: "Peterson, Jon" <jon.peterson@neustar.biz>
Cc: "'Conroy, Lawrence (SMTP)'" <lwc@roke.co.uk>, Rohan Mahy =
<rohan@cisco.com>,
      "'sipping@ietf.org'" <sipping@ietf.org>, bcampbell@dynamicsoft.com,
      "'Dean Willis'" <dean.willis@softarmor.com>, =
gonzalo.Camarillo@ericsson.com
Subject: RE: [Sipping] WGLC for draft-ietf-sipping-e164-02.txt
In-Reply-To: =
<0449D80A0E9B614A83FA9031B07E8D3B257A78@stntexch2.va.neustar.com>
References: =
<0449D80A0E9B614A83FA9031B07E8D3B257A78@stntexch2.va.neustar.com>
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Peterson, Jon writes:

  > I don't
  > think it is too much to ask that SIP applications utilizing ENUM be =
generous
  > in what they receive with regard to enumservices - it just doesn't =
seem
  > costly to be backwards compatible with the original iteration of the =
ENUM
  > standard.

check, for example, with cisco and let me know when they accept enum
replies with both formats. today they don't which is a very big
problem.=20

-- juha

15-May-2003 08:18:18 -0500,3663;000000000001-00000b10
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To: "Peterson, Jon" <jon.peterson@neustar.biz>
Cc: "'Conroy, Lawrence (SMTP)'" <lwc@roke.co.uk>, Rohan Mahy =
<rohan@cisco.com>,
      "'sipping@ietf.org'" <sipping@ietf.org>, bcampbell@dynamicsoft.com,
      "'Dean Willis'" <dean.willis@softarmor.com>, =
gonzalo.Camarillo@ericsson.com
Subject: RE: [Sipping] WGLC for draft-ietf-sipping-e164-02.txt
In-Reply-To: =
<0449D80A0E9B614A83FA9031B07E8D3B257A78@stntexch2.va.neustar.com>
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<0449D80A0E9B614A83FA9031B07E8D3B257A78@stntexch2.va.neustar.com>
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Peterson, Jon writes:

  > I don't
  > think it is too much to ask that SIP applications utilizing ENUM be =
generous
  > in what they receive with regard to enumservices - it just doesn't =
seem
  > costly to be backwards compatible with the original iteration of the =
ENUM
  > standard.

check, for example, with cisco and let me know when they accept enum
replies with both formats. today they don't which is a very big
problem.=20

-- juha
_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

15-May-2003 10:30:31 -0500,4982;000000000001-00000b11
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From: "Conroy, Lawrence (SMTP)" <lwc@roke.co.uk>
To: sipping@ietf.org
Cc: "Peterson, Jon" <jon.peterson@neustar.biz>,
      Juha Heinanen
                  <jh@lohi.eng.song.fi>
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Subject: [Sipping] e164 enumservice syntax support
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Hi Folks,
      [renamed the thread to be more specific]
* "Backward compatibility". There seem to be several choices
            here: (i) allow only RFC2916 syntax
                          (ii) allow only rfc2916bis syntax
                          (iii) allow both
            The e164-02 seems (apart from the add on in section 1 and
            section 7) to tend towards the first option (notably in the
            last sentence of section 5.1). However, with the inclusions
            of these caveats, it now seems to be sort of going for the
            last option.
            I object to this as it means we are producing an RFC that
            allows (or RECOMMENDs) syntax that we KNOW will be obsoleted
            in the very near future (unless someone needs to let me in
            on a secret :).
            I assume that -e164 is going to reference the 2916bis RFC,
            so it will be dependent on this - this isn't going to appear
            before 2916bis gets through the process. As part of the release
            of 2916bis, it obsoletes 2916 in the same way that 3403 obsoleted
            2915. Thus -e164 WILL be referring to obsoleted documents and
            syntax. The current text does not deprecate this syntax, so the
            -02 version of the draft (i.e the one we're reviewing) is at
            least adrift from "normal" IETF practice (see RFC2026 =
ss4.2.4,...).

            Now...given that 2916bis and the other enum drafts don't allow
            such backwards compatibility, why have it here?
            If this is a reference to RFC760 ss 3.2, then liberal means
            general flexibility with syntax; IMHO, -e164 does not need a
            specific reference to a (soon to be) Historic document.
            As Juha points out, this DOES cause confusion. As before, my
            vote's for option (ii).

all the best,
      Lawrence
--=20
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15-May-2003 10:32:23 -0500,6463;000000000001-00000b12
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Subject: RE: [Sipping] conf: comments on =
draft-rosenberg-sipping-conferencing-framework-01
Date: Thu, 15 May 2003 11:11:34 -0400
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Thread-Index: AcMKco2Qi9SqIWgYSg+Yf0l0fUHIuAQgNALQ
From: "Eric Burger" <eburger@snowshore.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: "Sipping (E-mail)" <sipping@ietf.org>
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My turn to apologize for being late. It's less than a month :-)

Most everything looks good. Some minor points in-line.

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Thu, April 24, 2003 11:00 AM
> To: Eric Burger
> Cc: Sipping (E-mail)
> Subject: Re: [Sipping] conf: comments on
> draft-rosenberg-sipping-conferencing-framework-01
>=20
>=20
> My apologies for the delay in responding. Thanks for your comments.
> Responses inline.
>=20
> Eric Burger wrote:

[snip]

> > Section 3.1, 4th & 5th paragraphs The example URI
> > ("discussion-on-dogs@example.com") would seem to contradict the
> > admonition that "a conference URI is never constructed or guessed =
by
> > a user." The key point is the conference URI is opaque. It could
> > mean something, or it could be a license plate. Requiring=20
> meaning is
> > BAD. Requiring obscuration is also BAD. Just leave it as opaque =
to
> > the framework.
>=20
> Having a URI with a mnemonic user part does not mean its no longer=20
> opaque to the system, no longer unique, or that the intent is=20
> for it to=20
> be guessed. For long lived conferences in particular, where=20
> there is a=20
> likelihood that someone will need to convey one over a phone, I see=20
> value in them being mnemonic.

I think we agree here. That is:
    1. There is value to having a mnemonic URI
    2. The URI is opaque to the protocol.

I was just pointing out that the text of the draft explicitly states =
the conference URI must not be a mnemonic (see quote above).

[snip]

> > Section 5.2.1 Is it worth mentioning a stateful application=20
> server as
> > a mechanism for running a conference?
>=20
> I don't understand. 5.2.1 is talking about adding users to a=20
> conference.=20
> What does that have to do with using an application server?

All INVITEs come from the application server. This means the =
application server is fully aware of the state of all call legs; can =
directly manipulate state, permissions, and media policy; and has the =
added bonus of making CPCP entirely unnecessary.

Given that the largest conference service bureaux in the world are on =
the order of a few tens of thousands of ports, I don't think the =
"application server doesn't scale like a proxy server" argument really =
holds.

[snip]

> > Section 5.4.2 and 5.4.3 I don't really get the difference between
> > these two examples.
>=20
> One of them uses CPCP to remove a user. This can be done by=20
> an automata.=20
> The other uses a web application to remove a user. THis can=20
> only be done=20
> by a user.
[snip]
> So, just because both SIP and CPCP can change the state of a=20
> conference,=20
> does not mean that they are the same protocol.
[snip]
> CPCP, being a data transaction protocol, would also allow you to=20
> directly query for the list. Just because there is this sliver of=20
> overlap does not mean that they are the same protocol.

A "sliver of overlap" is OK. A full set of overlap is not. Look at =
the draft, for just about every operation, the draft says, "Here is the =
way to do it with SIP, and here is the way to do it with CPCP." It may =
be my misreading, but it sounds like there is a lot of overlap.

[snip]

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Subject: [Sipping] status of sipping-e164-02.txt//enum-sip-00.txt
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Hi folks,
      [Renamed thread to be more specific]
I'm not sure what the relationship is between this I-D and the
enum-sip I-D. I had wondered if the sipping- draft was going to
be superceded by the enum- draft, but there is a lot of useful
exemplary information in the sipping- draft that gives good
guidance to implementors. However, there are chunks of text in
this one that seem appropriate for a 'sip' enumservice definition
(e.g. section 5). Thus, what is the role for this draft; is it
informational, or standards-track, and if so, what is it defining,
given that there's also an enum-sip draft on the standards track?

=3D> IMHO, the sipping- draft does need to be re-addressed in light
of the 'sip' enumservices draft. At present there are overlaps in
several of the sections.


There are some more detailed issues, of course; notably in the
treatment of order and preference fields in section 6.3 - time has
moved on since this draft was issued, and it will need to be changed.

Also, as an aside, some of the statements that seem to mandate a
particular treatment do not include the usual SHALL, MUST, ..keywords.
For example, the penultimate sentence in the 2nd paragraph of section
5.3, which is an instruction, but does not have MUST (only 'is =
always').
This is followed immediately by a sentence that has a SHOULD. Choose.

This draft needs an up-issue, IMHO, and a decision on whether it is
standards-track or informational (it is silent on the matter).

all the best,
      Lawrence
--=20
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_______________________________________________
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14-May-2003 03:45:10 -0500,1777;000000000000-00000b14
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From: "Gonzalo Camarillo Gonzalez (LMF)"
                  <Gonzalo.Camarillo@lmf.ericsson.se>
To: "'sipping@ietf.org'" <sipping@ietf.org>
Cc: "Jon Peterson (E-mail)" <jon.peterson@neustar.biz>,
      "Rohan Mahy (E-mail)" <rohan@cisco.com>,
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                  <dean.willis@softarmor.com>
Subject: Last Call for early media use cases
Date: Wed, 14 May 2003 09:12:50 +0200
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Folks,

in the last IETF meeting, we requested early media use cases that show =
the need of a "media coming" flag. So far, we have not received =
anything.

We intend to release a new version of the early media draft in a week =
or so. So, that is the time you have to send me your use cases.

Thanks,

Gonzalo

13-May-2003 23:02:49 -0500,1682;000000000001-00000b15
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Date: Tue, 13 May 2003 21:02:01 -0700
Subject: WGLC for draft-ietf-sipping-e164-02.txt
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gonzalo.Camarillo@ericsson.com
To: "'sipping@ietf.org'" <sipping@ietf.org>
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Hello Everyone,

I'd like to begin Working Group Last Call on:

http:/www.ietf.org/internet-drafts/draft-ietf-sipping-e164-02.txt

this WGLC will end on June 13, 2003.

thanks,
-rohan

15-May-2003 13:40:44 -0500,6996;000000000001-00000b16
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Conroy, Lawrence (SMTP)'" <lwc@roke.co.uk>, sipping@ietf.org
Cc: Juha Heinanen <jh@lohi.eng.song.fi>
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Subject: [Sipping] RE: e164 enumservice syntax support
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I think we both agree that e164-02 specifies (iii), and I think e164-03
should as well. RF2026 does not mandate, as you seem to suggest, that
behavior for backwards compatibility is illegal in IETF standards. ENUM =
made
a decision (the reasons for which are outside the scope of our work in
SIPPING) to reverse its enumservice string format in order to break
automatic backwards compatibility, yes. This does not change the fact =
that
several ENUM implementations have been constructed in the three years =
since
the publication of RFC2916 that use the service field format of =
RFC2916.
Some implementations, no doubt, use those formats today.

There are numerous RFCs that contain guidance for dealing with
implementations that are compliant with older versions of a =
specification
(for example, the SIP RFC, RFC3261, contains numerous references to =
RFC2543
and quite a bit of behavior that is intended to foster backwards
compatibility) - this is in no way suspect or broken from a process
perspective. I don't think that ENUM clients acting on behalf of SIP
applications which retrieve usable resource records should be forced to
discard these records because the enumservice uses a standard format =
that
has been superannuated. Considering that there are probably =
implementations
out there in the field today that will return such records, I think =
that
would be a poor decision. I don't think any "confusion" that might =
result
from this has yet been substantiated. Nor do I think it is an unseemly
burden for implementers to support two variations on a seven-character
string instead of one.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Conroy, Lawrence (SMTP) [mailto:lwc@roke.co.uk]
> Sent: Thursday, May 15, 2003 8:19 AM
> To: sipping@ietf.org
> Cc: Peterson, Jon; Juha Heinanen
> Subject: e164 enumservice syntax support
>=20
>=20
> Hi Folks,
> [renamed the thread to be more specific]
> * "Backward compatibility". There seem to be several choices
> here: (i) allow only RFC2916 syntax
> (ii) allow only rfc2916bis syntax
> (iii) allow both
> The e164-02 seems (apart from the add on in section 1 and
> section 7) to tend towards the first option (notably in the
> last sentence of section 5.1). However, with the inclusions
> of these caveats, it now seems to be sort of going for the
> last option.
> I object to this as it means we are producing an RFC that
> allows (or RECOMMENDs) syntax that we KNOW will be obsoleted
> in the very near future (unless someone needs to let me in
> on a secret :).
> I assume that -e164 is going to reference the 2916bis RFC,
> so it will be dependent on this - this isn't going to appear
> before 2916bis gets through the process. As part of the release
> of 2916bis, it obsoletes 2916 in the same way that 3403 =
obsoleted
> 2915. Thus -e164 WILL be referring to obsoleted documents and
> syntax. The current text does not deprecate this syntax, so the
> -02 version of the draft (i.e the one we're reviewing) is at
> least adrift from "normal" IETF practice (see RFC2026 =
ss4.2.4,...).
>=20
> Now...given that 2916bis and the other enum drafts don't allow
> such backwards compatibility, why have it here?
> If this is a reference to RFC760 ss 3.2, then liberal means
> general flexibility with syntax; IMHO, -e164 does not need a
> specific reference to a (soon to be) Historic document.
> As Juha points out, this DOES cause confusion. As before, my
> vote's for option (ii).
>=20
> all the best,
> Lawrence
> --=20
> --------------------------------------------------------------
> ---------
> Roke Manor Research : This information is provided "as is"=20
> and is not
> <mailto:lwc@roke.co.uk>: intended to create any contractual or legal
> <tel:+441794833666> : relationship.
>=20
_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Conroy, Lawrence (SMTP)'" <lwc@roke.co.uk>, sipping@ietf.org
Cc: Rohan.Mahy@cisco.com
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Subject: [Sipping] RE: status of sipping-e164-02.txt//enum-sip-00.txt
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Unlike draft-ietf-enum-sip, draft-ietf-sipping-e164 exists to describe =
how
SIP implementations should handle E.164 numbers, which may appear in
contexts in which ENUM is impracticable or uninformative (such as =
gateway
location scenarios, cases in which there is no ENUM record =
corresponding to
a particular E.164 number, etc). As you may have noted, it is not an
enumservice definition, but it references and uses an enumservice =
definition
(see sipping-e164 reference [15]). While there are some overlaps =
between the
two drafts (draft-ietf-enum-sip, especially in Section 3, summarizes =
some of
the conclusions of sipping-e164), I don't feel these damage or devalue
either of the documents. That much said, as I revise this document in
response to last call comments, I'll try to seek out and remove any
unhelpful redundancy. Redundancy with rfc2916bis, which has become =
better
specified over time, is probably more likely, and I'll also keep an eye =
out
for that.

>=20
> There are some more detailed issues, of course; notably in the
> treatment of order and preference fields in section 6.3 - time has
> moved on since this draft was issued, and it will need to be changed.

Um, while I don't think the behaviorial recommendations would really =
change
(i.e. use one order when you author ENUM records and differentiate by
preference), I'd agree that 6.3 isn't so crucial, given that rfc2916bis =
now
has section 1.3. I could make 6.3 much shorter and less specific today;
prior versions of rfc2916bis had much less guidance about dealing with
order.

> Also, as an aside, some of the statements that seem to mandate a
> particular treatment do not include the usual SHALL, MUST, =
..keywords.
> For example, the penultimate sentence in the 2nd paragraph of section
> 5.3, which is an instruction, but does not have MUST (only 'is =
always').
> This is followed immediately by a sentence that has a SHOULD. Choose.
>=20

I think the "is always" in 5.3 is not an instruction - it's an =
observation.
The sentence following it is a normative instruction. Really, the "is
always" sentence asserted only that in RFC2916 (and in bis), this "is
always" how examples were constructed. That could be clearer, though, =
sure,
and it will be.

> This draft needs an up-issue, IMHO, and a decision on whether it is
> standards-track or informational (it is silent on the matter).

As I said, I'll be issuing a new revision, and I'll incorporate the =
comments
above. However, IETF documents do not declare their intended status, at
least as far as I know. sipping-e164 is not meant to be Informational - =
it
contains numerous normative guidelines, and was intended for PS.=20

Jon Peterson
NeuStar, Inc.

> all the best,
> Lawrence
> --=20
> --------------------------------------------------------------
> ---------
> Roke Manor Research : This information is provided "as is"=20
> and is not
> <mailto:lwc@roke.co.uk>: intended to create any contractual or legal
> <tel:+441794833666> : relationship.
>=20
_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

15-May-2003 19:57:05 -0500,6437;000000000001-00000b18
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Conroy, Lawrence (SMTP)'" <lwc@roke.co.uk>,
      "'sipping@ietf.org'"
                  <sipping@ietf.org>
Cc: "'Peterson, Jon'" <jon.peterson@neustar.biz>,
      "'Juha Heinanen'"
                  <jh@lohi.eng.song.fi>
Subject: RE: [Sipping] e164 enumservice syntax support
Date: Fri, 16 May 2003 01:45:26 +0100
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Perhaps an appropriate AD can make a definitive statement on the future =
of RFC 2916.

It would appear to me that it is entirely inappropriate to have a =
reference in a new RFC to an RFC that is already obsolete. So is this =
the situation that will exist?

This is not the same case as when a revised RFC refers to the previous =
version of the same RFC. The appropriate place to cover that case is in =
rfc2916bis with respect to RFC 2916, rather than in the sipping e164 =
document. I understand that it does not do that.

regards

Keith

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com=20

> -----Original Message-----
> From: Conroy, Lawrence (SMTP) [mailto:lwc@roke.co.uk]
> Sent: 15 May 2003 16:19
> To: sipping@ietf.org
> Cc: Peterson, Jon; Juha Heinanen
> Subject: [Sipping] e164 enumservice syntax support
>=20
>=20
> Hi Folks,
> [renamed the thread to be more specific]
> * "Backward compatibility". There seem to be several choices
> here: (i) allow only RFC2916 syntax
> (ii) allow only rfc2916bis syntax
> (iii) allow both
> The e164-02 seems (apart from the add on in section 1 and
> section 7) to tend towards the first option (notably in the
> last sentence of section 5.1). However, with the inclusions
> of these caveats, it now seems to be sort of going for the
> last option.
> I object to this as it means we are producing an RFC that
> allows (or RECOMMENDs) syntax that we KNOW will be obsoleted
> in the very near future (unless someone needs to let me in
> on a secret :).
> I assume that -e164 is going to reference the 2916bis RFC,
> so it will be dependent on this - this isn't going to appear
> before 2916bis gets through the process. As part of the release
> of 2916bis, it obsoletes 2916 in the same way that 3403=20
> obsoleted
> 2915. Thus -e164 WILL be referring to obsoleted documents and
> syntax. The current text does not deprecate this syntax, so the
> -02 version of the draft (i.e the one we're reviewing) is at
> least adrift from "normal" IETF practice (see RFC2026=20
> ss4.2.4,...).
>=20
> Now...given that 2916bis and the other enum drafts don't allow
> such backwards compatibility, why have it here?
> If this is a reference to RFC760 ss 3.2, then liberal means
> general flexibility with syntax; IMHO, -e164 does not need a
> specific reference to a (soon to be) Historic document.
> As Juha points out, this DOES cause confusion. As before, my
> vote's for option (ii).
>=20
> all the best,
> Lawrence
> --=20
> --------------------------------------------------------------
> ---------
> Roke Manor Research : This information is provided "as is"=20
> and is not
> <mailto:lwc@roke.co.uk>: intended to create any contractual or legal
> <tel:+441794833666> : relationship.
> _______________________________________________
> Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20
_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

16-May-2003 04:18:50 -0500,3560;000000000001-00000b19
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From: Nicole Johnson <NJ2@dataconnection.com>
To: "'sipping@ietf.org'" <sipping@ietf.org>
Date: Fri, 16 May 2003 10:03:43 +0100
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Subject: [Sipping] [Sipping]Questions on =
draft-ietf-sipping-connect-reuse-reqs-00
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I have a couple of questions about =
draft-ietf-sipping-connect-reuse-reqs-00.

1. Intended scope of alias parameter. =20

Is the aim of this draft to enable connections to be reused within a =
single
dialog, or across multiple dialogs, i.e. can an alias learned within =
the
scope of one dialog be used in another? Is the alias parameter valid =
on
out-of-dialog requests?

2. Target address fails to authenticate.

If a server receives a request containing the alias parameter and the =
target
address of the request fails authentication, how should the server =
react?
It could drop the message, respond to the target address with a "not
authorized" response, or continue processing the request as if the =
"alias"
parameter was not present. Is there a recommended behaviour?=20

Thanks,
Nicole
_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

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From: David Schwartz <davids@deltathree.com>
To: "'rohan@cisco.com'" <rohan@cisco.com>
Cc: "'Nachumf@audiocodes.com'" <Nachumf@audiocodes.com>,
      "'sip_developer@hotmail.com'" <sip_developer@hotmail.com>,
      "'sipping@ietf.org'" <sipping@ietf.org>,
      "'AttilaS@vegastream.com'"
                  <AttilaS@vegastream.com>,
      "'sip-implementors@cs.columbia.edu'"
                  <sip-implementors@cs.columbia.edu>,
      "'rizsanyi@neobee.net'"
                  <rizsanyi@neobee.net>
Date: Wed, 14 May 2003 11:05:18 +0300
X-Mailer: Internet Mail Service (5.5.2653.19)
Subject: [Sipping] Re: [Sip-implementers] dtmf digits using INFO ... =
SIPauthors ...
  Can you answer please ?
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Hi Rohan.

I would like to add to Nachum's comment.

Here at deltathree we have been using an IVR type system for about two =
years
now with originating sip devices contacting our network from many =
different
locations. For termination we are using Cisco Gateways that address the
double digit issue (as Nachum stated). What we have found is that in
locations that have very poor infrastructure (usually satellite) =
RFC2833 is
highly unreliable. In these locations where latency and packet loss are
high, and the traffic may be bursty in nature, the retransmission =
mechanism
associated with INFO requests assures the arrival of all the digits =
whereas
2833 is not providing us with the same completion rates.=20

When entering a PIN code for instance where the loss of even one digit =
is
destructive we have had higher success rates with INFO type solutions
(vegastream, audiocodes, etc,) than we have had with pure RFC2833 =
devices.
True there is a longer delay associated with this approach as each =
digit is
sent only after receiving a 200 response but the alternative is in many
cases not being able to enter the PIN at all.

David Schwartz

-----Original Message-----
From: sipping-request@ietf.org [mailto:sipping-request@ietf.org]
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To: sipping@ietf.org
Subject: Sipping digest, Vol 1 #1017 - 4 msgs


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Today's Topics:

      1. RE: Re: [Sip-implementors] dtmf digits using INFO ... SIPauthors =
...
Can you answer please ? (Nachum Frid)
      2. Re: Re: [Sip-implementors] dtmf digits using INFO ... SIPauthors =
...
Can you answer please ? (Rohan Mahy)
      3. RE: Re: [Sip-implementors] dtmf digits using INFO ... S
              IPauthors ... Can you answer please ? (Francois Audet)

--__--__--

Message: 1
Subject: RE: [Sipping] Re: [Sip-implementors] dtmf digits using INFO =
...
SIPauthors ... Can you answer please ?
Date: Tue, 13 May 2003 19:42:06 +0200
From: "Nachum Frid" <Nachumf@audiocodes.com>
To: "Rohan Mahy" <rohan@cisco.com>,
      "SIP Developer" <sip_developer@hotmail.com>
Cc: <sipping@ietf.org>, "Attila Sipos" <AttilaS@vegastream.com>,
      <sip-implementors@cs.columbia.edu>, <rizsanyi@neobee.net>

              Rohan,
            =20
              In your draft you are describing the double DTMF problem:
              "
              Under this proposal, a User Agent MUST NOT send signaled digits =
or
                    telephone-events using the INFO method if the event was ever
                    represented as a tone (as media). Only signals originated as =
pure
                    signaling MAY generate an INFO method. Failure to heed this
                    requirement will result in double-detection of digits/events
            =20
              If INFO is used incorrectly by a pair of PSTN gateways (for =
example),
              the source gateway may detect a digit, send an INFO request =
which is
lost, and retransmit that request. The target gateway would send the
              original in-band tone to the PSTN when the audio media arrives, =
later
              when the INFO arrives, the target gateway would render the same =
tone
              again. It is quite likely that the INFO will arrive after the =
media
              tone has been played (especially if multiple intermediaries are
              involved); the same digit would be played out again and this =
will
              frequently cause systems in the PSTN to detect the same digit
              signaled twice."
            =20
            =20
            =20
              Most of the PSTN gateways we are familiar with (Cisco, =
AudioCodes)
will mute the DTMF signal, when they are configured to use out of band
DTMF(H.245 alphanumeric, or SIP INFO). Usually source gateway will =
detect
DTMF digit coming from the PSTN side and erase it from the voice stream =
not
relaying it to IP. Than, the INFO (or H.245) message is send to the =
target
gateway, which will regenerate the DTMF to the PSTN side, without =
playing
double digits. In H.323 it works already for few years.
            =20
              Nachum
            =20
            =20
              -----Original Message-----
From: Rohan Mahy [mailto:rohan@cisco.com]=20
Sent: Tuesday, May 13, 2003 1:34 AM
To: SIP Developer
Cc: sipping@ietf.org; Attila Sipos; sip-implementors@cs.columbia.edu;
rizsanyi@neobee.net
Subject: Re: [Sipping] Re: [Sip-implementors] dtmf digits using INFO =
...
SIPauthors ... Can you answer please ?
            =20
              Hi Rohan,
            =20
              The main reason to not carry DTMF in SIP INFOs is that there is =
a
very =20
              good chance that you will get double-digits or otherwise =
corrupted =20
              digits because the signaling is not time-synchronized with the =
audio

              media. This issue is discussed in an expired draft which I =
wrote a =20
              long time ago, which I abandoned in favor of the work generated =
by
the =20
              app-interaction design team. At this point my draft is useful =
only in

              that it explains how double digits tend to occur.
            =20
              my expired draft:
            =20
              =
http://www.softarmor.com/sipping/drafts/morgue/draft-mahy-sipping-=20
              signaled-digits-01.html
            =20
              the app-interaction design team drafts:
            =20
=20
http://www.softarmor.com/sipping/drafts/draft-burger-sipping-kpml-01.txt=

              =
http://www.softarmor.com/sipping/drafts/draft-culpepper-sipping-app-=20
              interact-reqs-03.txt
              =
http://www.softarmor.com/sipping/drafts/draft-jennings-sip-app-info-=20
              00.txt
              =
http://www.softarmor.com/sipping/drafts/draft-rosenberg-sipping-app-=20
              interaction-framework-00.txt
            =20
            =20
              thanks,
              -rohan (Mahy)
            =20
            =20
              On Friday, May 9, 2003, at 11:05 AM, SIP Developer wrote:
            =20
              > Hi All,
              >
              > I hate to say that but SIP community doesn't agree
              > or think that there are lots of use and requirement for =
carrying
              > DTMF Out-of-Band.
              >
              > Every month, I can see that someone asks the same question =
again
and =20
              > again
              > that how do we send DTMF via INFO and Out-Of-Band .
              >
              > 3pcc needs it, lots of service creation environment needs it =
and
              > doesn't it looks promising to have a way in SIP to do OOB
              > DTMF just like h245-alphanumeric in H.323.
              >
              > If we think of applications, then there are lots of them that =
I
can
              > count which doesn't care about the timing info carried by =
RFC2833.
              > And if we really care about timing, then carrying RFC2833 via
              > notify (Again OOB) is promising.
              >
              > I really do not understand if this is a political issue to not
promote
              > SIP INFO for carrying DTMF or if someone can justify it to me
              > that there is no use to carry DTMF OOB.
              >
              > SIP authors, can you answer why SIP INFO for carrying simple
              > DTMF is not a standard solution ?
              >
              > Thanks,
              > Rohan Kumar
              >
              >
              > ----- Original Message -----
              > From: "Attila Sipos" <AttilaS@vegastream.com>
              > To: <rizsanyi@neobee.net>; <sip-implementors@cs.columbia.edu>
              > Sent: Friday, May 09, 2003 9:50 AM
              > Subject: RE: [Sip-implementors] dtmf digits using INFO
              >
              >
              >>
              >> Hello Zsolt,
              >>
              >> I still don't think there is a fully-agreed
              >> standard. However, as much as I hate to say it,
              >> CISCO have their own proposed standard which some
              >> people (like us) interoperate with:
              >>
              >>
              > =
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/=20
              > 122newft/122
              >> t/122t11/ftinfo.htm
              >>
              >> Regards,
              >>
              >> Attila
              >>
              >> Attila Sipos
              >> Software Engineer
              >> <http://www.vegastream.com>
              >> VegaStream : A World of difference for your Integrated
Communications
              >>
              >> (jo napot kiv=E1nok)
              >>
              >>
              >>
              >>> -----Original Message-----
              >>> From: Rizsanyi Zsolt [mailto:rizsanyi@myrealbox.com]
              >>> Sent: 09 May 2003 13:41
              >>> To: sip-implementors@cs.columbia.edu
              >>> Subject: [Sip-implementors] dtmf digits using INFO
              >>>
              >>>
              >>> Hi!
              >>>
              >>> Could somebody refer me to a current standard or draft, on
              >>> how to send dtmf
              >>> digits using the INFO method?
              >>>
              >>> Thanks
              >>> Zsolt Rizsanyi
              >>>
              >>> _______________________________________________
              >>> Sip-implementors mailing list
              >>> Sip-implementors@cs.columbia.edu
              >>> =
http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
              >>>
              >>
              >> _______________________________________________
              >> Sip-implementors mailing list
              >> Sip-implementors@cs.columbia.edu
              >> =
http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
              >>
              > _______________________________________________
              > Sipping mailing list
https://www1.ietf.org/mailman/listinfo/sipping
              > This list is for NEW development of the application of SIP
              > Use sip-implementors@cs.columbia.edu for questions on current =
sip
              > Use sip@ietf.org for new developments of core SIP
            =20
            =20
              _______________________________________________
              Sip-implementors mailing list
              Sip-implementors@cs.columbia.edu
              http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors

--__--__--

Message: 2
Date: Tue, 13 May 2003 09:56:06 -0700
Subject: Re: [Sipping] Re: [Sip-implementors] dtmf digits using INFO =
...
SIPauthors ... Can you answer please ?
Cc: "SIP Developer" <sip_developer@hotmail.com>, <sipping@ietf.org>,
      "Attila Sipos" <AttilaS@vegastream.com>,
<sip-implementors@cs.columbia.edu>,
      <rizsanyi@neobee.net>
To: "Nachum Frid" <Nachumf@audiocodes.com>
From: Rohan Mahy <rohan@cisco.com>


On Tuesday, May 13, 2003, at 10:42 AM, Nachum Frid wrote:

> Rohan,
>
> In your draft you are describing the double DTMF problem:
> "
> Under this proposal, a User Agent MUST NOT send signaled =
digits =20
> or
> telephone-events using the INFO method if the event was =
ever
> represented as a tone (as media). Only signals originated =

> as pure
> signaling MAY generate an INFO method. Failure to heed =
this
> requirement will result in double-detection of =
digits/events
>
> If INFO is used incorrectly by a pair of PSTN gateways (for =20
> example),
> the source gateway may detect a digit, send an INFO request =20
> which is lost, and retransmit that request. The target gateway =
would

> send the
> original in-band tone to the PSTN when the audio media =
arrives, =20
> later
> when the INFO arrives, the target gateway would render the =
same =20
> tone
> again. It is quite likely that the INFO will arrive after the =
=20
> media
> tone has been played (especially if multiple intermediaries =
are
> involved); the same digit would be played out again and this =20
> will
> frequently cause systems in the PSTN to detect the same digit
> signaled twice."
>
>
>
> Most of the PSTN gateways we are familiar with (Cisco, =20
> AudioCodes) will mute the DTMF signal, when they are configured to =
use =20
> out of band DTMF(H.245 alphanumeric, or SIP INFO). Usually source =20
> gateway will detect DTMF digit coming from the PSTN side and erase it =
=20
> from the voice stream not relaying it to IP.

In my experience the muting/erasure is imprecise and imperfect at best. =
=20
    Frequently, by the time accurate detection occurs, a bit of tone has =

already leaked out. If you tune your DSP to be more aggressive at =20
removing tones, then you can get false positives and mute harmonics of =

speech. I could go on, but I think you get the idea. i have seen these =
=20
problems with Cisco H.323 gateways and AudioCodes gateways that a Cisco =
=20
aquisition OEMed, so you can't convince me the problem doesn't exist.

> Than, the INFO (or H.245) message is send to the target gateway, =20
> which will regenerate the DTMF to the PSTN side, without playing =20
> double digits. In H.323 it works already for few years.

It works poorly enough that the ITU added support for RFC 2833 in H.323 =
=20
v4 to combat this problem.

thanks,
-rohan


> Nachum
>
>
> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@cisco.com]
> Sent: Tuesday, May 13, 2003 1:34 AM
> To: SIP Developer
> Cc: sipping@ietf.org; Attila Sipos; sip-implementors@cs.columbia.edu; =
=20
> rizsanyi@neobee.net
> Subject: Re: [Sipping] Re: [Sip-implementors] dtmf digits using INFO =

> ... SIPauthors ... Can you answer please ?
>
> Hi Rohan,
>
> The main reason to not carry DTMF in SIP INFOs is that there =
is =20
> a very
> good chance that you will get double-digits or otherwise =20
> corrupted
> digits because the signaling is not time-synchronized with the =
=20
> audio
> media. This issue is discussed in an expired draft which I =20
> wrote a
> long time ago, which I abandoned in favor of the work =
generated =20
> by the
> app-interaction design team. At this point my draft is useful =

> only in
> that it explains how double digits tend to occur.
>
> my expired draft:
>
> =20
> http://www.softarmor.com/sipping/drafts/morgue/draft-mahy-sipping-
> signaled-digits-01.html
>
> the app-interaction design team drafts:
>
> =20
> http://www.softarmor.com/sipping/drafts/draft-burger-sipping-kpml-=20
> 01.txt
> =20
> http://www.softarmor.com/sipping/drafts/draft-culpepper-sipping-app-
> interact-reqs-03.txt
> =20
> http://www.softarmor.com/sipping/drafts/draft-jennings-sip-app-info-
> 00.txt
> =20
> http://www.softarmor.com/sipping/drafts/draft-rosenberg-sipping-app-
> interaction-framework-00.txt
>
>
> thanks,
> -rohan (Mahy)
>
>
> On Friday, May 9, 2003, at 11:05 AM, SIP Developer wrote:
>
>> Hi All,
>>
>> I hate to say that but SIP community doesn't agree
>> or think that there are lots of use and requirement for carrying
>> DTMF Out-of-Band.
>>
>> Every month, I can see that someone asks the same question again and
>> again
>> that how do we send DTMF via INFO and Out-Of-Band .
>>
>> 3pcc needs it, lots of service creation environment needs it and
>> doesn't it looks promising to have a way in SIP to do OOB
>> DTMF just like h245-alphanumeric in H.323.
>>
>> If we think of applications, then there are lots of them that I can
>> count which doesn't care about the timing info carried by RFC2833.
>> And if we really care about timing, then carrying RFC2833 via
>> notify (Again OOB) is promising.
>>
>> I really do not understand if this is a political issue to not =
promote
>> SIP INFO for carrying DTMF or if someone can justify it to me
>> that there is no use to carry DTMF OOB.
>>
>> SIP authors, can you answer why SIP INFO for carrying simple
>> DTMF is not a standard solution ?
>>
>> Thanks,
>> Rohan Kumar
>>
>>
>> ----- Original Message -----
>> From: "Attila Sipos" <AttilaS@vegastream.com>
>> To: <rizsanyi@neobee.net>; <sip-implementors@cs.columbia.edu>
>> Sent: Friday, May 09, 2003 9:50 AM
>> Subject: RE: [Sip-implementors] dtmf digits using INFO
>>
>>
>>>
>>> Hello Zsolt,
>>>
>>> I still don't think there is a fully-agreed
>>> standard. However, as much as I hate to say it,
>>> CISCO have their own proposed standard which some
>>> people (like us) interoperate with:
>>>
>>>
>> http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/
>> 122newft/122
>>> t/122t11/ftinfo.htm
>>>
>>> Regards,
>>>
>>> Attila
>>>
>>> Attila Sipos
>>> Software Engineer
>>> <http://www.vegastream.com>
>>> VegaStream : A World of difference for your Integrated =
Communications
>>>
>>> (jo napot kiv=E1nok)
>>>
>>>
>>>
>>>> -----Original Message-----
>>>> From: Rizsanyi Zsolt [mailto:rizsanyi@myrealbox.com]
>>>> Sent: 09 May 2003 13:41
>>>> To: sip-implementors@cs.columbia.edu
>>>> Subject: [Sip-implementors] dtmf digits using INFO
>>>>
>>>>
>>>> Hi!
>>>>
>>>> Could somebody refer me to a current standard or draft, on
>>>> how to send dtmf
>>>> digits using the INFO method?
>>>>
>>>> Thanks
>>>> Zsolt Rizsanyi
>>>>
>>>> _______________________________________________
>>>> Sip-implementors mailing list
>>>> Sip-implementors@cs.columbia.edu
>>>> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
>>>>
>>>
>>> _______________________________________________
>>> Sip-implementors mailing list
>>> Sip-implementors@cs.columbia.edu
>>> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
>>>
>> _______________________________________________
>> Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
>
>
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@cs.columbia.edu
> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors


--__--__--

Message: 3
From: "Francois Audet" <audet@nortelnetworks.com>
To: "'Rohan Mahy'" <rohan@cisco.com>, "'Nachum Frid'"
<Nachumf@audiocodes.com>
Cc: "'SIP Developer'" <sip_developer@hotmail.com>,
      "'sipping@ietf.org'"
                  <sipping@ietf.org>,
      "'Attila Sipos'" <AttilaS@vegastream.com>,
      "'sip-implementors@cs.columbia.edu'" =
<sip-implementors@cs.columbia.edu>,
      "'rizsanyi@neobee.net'" <rizsanyi@neobee.net>
Subject: RE: [Sipping] Re: [Sip-implementors] dtmf digits using INFO =
... S
                IPauthors ... Can you answer please ?
Date: Tue, 13 May 2003 11:20:30 -0700

This message is in MIME format. Since your mail reader does not =
understand
this format, some or all of this message may not be legible.

------_=3D_NextPart_001_01C3197C.55FF4E20
The main reason for adding RFC2833 support in H.323v4 (despite the fact t=
he
earlier version supported long tones out-of-band) were:

- Interoperability with systems that use RFC2833. (That means for example
SIP-to-H.323)

- IVR/Call center applications where time synchronization is necessary.
(This applies to SIP as well of course). For example, you could have a
"menu" item in a voice application where pressing a digit will cause a
recording to stop. You could end on somebody's mailbox, where pressing "9=
"
would terminate your recording. If you press 9 and the recording is stopp=
ed
a couple of second later because the signalling message took longer to
arrive, you could unwillingly record something you didn't want to record
(like an insult).=20

Of course this doesn't mean that you'd always need RFC2833 in-band (the
proof is that H.323 systems have been running for years), but there are
indeed difficulties with the current VoIP systems and IVR/Call
Center/Voicemail.

The main problem is the transistion from out-of-band to RFC2833.

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@cisco.com]=20
> Sent: Tuesday, May 13, 2003 09:56
> To: Nachum Frid
> Cc: SIP Developer; sipping@ietf.org; Attila Sipos;=20
> sip-implementors@cs.columbia.edu; rizsanyi@neobee.net
> Subject: Re: [Sipping] Re: [Sip-implementors] dtmf digits=20
> using INFO ... SIPauthors ... Can you answer please ?
>=20
>=20
>=20
> On Tuesday, May 13, 2003, at 10:42 AM, Nachum Frid wrote:
>=20
> > Rohan,
> >
> > In your draft you are describing the double DTMF problem:
> > "
> > Under this proposal, a User Agent MUST NOT send=20
> signaled digits
> > or
> > telephone-events using the INFO method if the=20
> event was ever
> > represented as a tone (as media). Only signals=20
> originated =20
> > as pure
> > signaling MAY generate an INFO method. Failure=20
> to heed this
> > requirement will result in double-detection of=20
> digits/events
> >
> > If INFO is used incorrectly by a pair of PSTN gateways (for
> > example),
> > the source gateway may detect a digit, send an INFO request =20
> > which is lost, and retransmit that request. The target=20
> gateway would =20
> > send the
> > original in-band tone to the PSTN when the audio=20
> media arrives, =20
> > later
> > when the INFO arrives, the target gateway would=20
> render the same =20
> > tone
> > again. It is quite likely that the INFO will arrive=20
> after the =20
> > media
> > tone has been played (especially if multiple=20
> intermediaries are
> > involved); the same digit would be played out again=20
> and this =20
> > will
> > frequently cause systems in the PSTN to detect the same digit
> > signaled twice."
> >
> >
> >
> > Most of the PSTN gateways we are familiar with (Cisco, =20
> > AudioCodes) will mute the DTMF signal, when they are=20
> configured to use =20
> > out of band DTMF(H.245 alphanumeric, or SIP INFO). Usually=20
> source =20
> > gateway will detect DTMF digit coming from the PSTN side=20
> and erase it =20
> > from the voice stream not relaying it to IP.
>=20
> In my experience the muting/erasure is imprecise and=20
> imperfect at best. =20
> Frequently, by the time accurate detection occurs, a bit of=20
> tone has =20
> already leaked out. If you tune your DSP to be more aggressive at =20
> removing tones, then you can get false positives and mute=20
> harmonics of =20
> speech. I could go on, but I think you get the idea. i have=20
> seen these =20
> problems with Cisco H.323 gateways and AudioCodes gateways=20
> that a Cisco =20
> aquisition OEMed, so you can't convince me the problem doesn't exist.
>=20
> > Than, the INFO (or H.245) message is send to the target gateway, =20
> > which will regenerate the DTMF to the PSTN side, without playing =20
> > double digits. In H.323 it works already for few years.
>=20
> It works poorly enough that the ITU added support for RFC=20
> 2833 in H.323 =20
> v4 to combat this problem.
>=20
> thanks,
> -rohan
>=20
>=20
> > Nachum
> >
> >
> > -----Original Message-----
> > From: Rohan Mahy [mailto:rohan@cisco.com]
> > Sent: Tuesday, May 13, 2003 1:34 AM
> > To: SIP Developer
> > Cc: sipping@ietf.org; Attila Sipos;=20
> sip-implementors@cs.columbia.edu; =20
> > rizsanyi@neobee.net
> > Subject: Re: [Sipping] Re: [Sip-implementors] dtmf digits=20
> using INFO =20
> > ... SIPauthors ... Can you answer please ?
> >
> > Hi Rohan,
> >
> > The main reason to not carry DTMF in SIP INFOs is=20
> that there is =20
> > a very
> > good chance that you will get double-digits or otherwise =20
> > corrupted
> > digits because the signaling is not=20
> time-synchronized with the =20
> > audio
> > media. This issue is discussed in an expired draft which I =20
> > wrote a
> > long time ago, which I abandoned in favor of the=20
> work generated =20
> > by the
> > app-interaction design team. At this point my draft=20
> is useful =20
> > only in
> > that it explains how double digits tend to occur.
> >
> > my expired draft:
> >
> > =20
> > http://www.softarmor.com/sipping/drafts/morgue/draft-mahy-sipping-
> > signaled-digits-01.html
> >
> > the app-interaction design team drafts:
> >
> > =20
> > http://www.softarmor.com/sipping/drafts/draft-burger-sipping-kpml-=20
> > 01.txt
> > =20
> > http://www.softarmor.com/sipping/drafts/draft-culpepper-sipping-app-
> > interact-reqs-03.txt
> > =20
> > http://www.softarmor.com/sipping/drafts/draft-jennings-sip-app-info-
> > 00.txt
> > =20
> > http://www.softarmor.com/sipping/drafts/draft-rosenberg-sipping-app-
> > interaction-framework-00.txt
> >
> >
> > thanks,
> > -rohan (Mahy)
> >
> >
> > On Friday, May 9, 2003, at 11:05 AM, SIP Developer wrote:
> >
> >> Hi All,
> >>
> >> I hate to say that but SIP community doesn't agree
> >> or think that there are lots of use and requirement for carrying
> >> DTMF Out-of-Band.
> >>
> >> Every month, I can see that someone asks the same question=20
> again and
> >> again
> >> that how do we send DTMF via INFO and Out-Of-Band .
> >>
> >> 3pcc needs it, lots of service creation environment needs it and
> >> doesn't it looks promising to have a way in SIP to do OOB
> >> DTMF just like h245-alphanumeric in H.323.
> >>
> >> If we think of applications, then there are lots of them=20
> that I can
> >> count which doesn't care about the timing info carried by RFC2833.
> >> And if we really care about timing, then carrying RFC2833 via
> >> notify (Again OOB) is promising.
> >>
> >> I really do not understand if this is a political issue to=20
> not promote
> >> SIP INFO for carrying DTMF or if someone can justify it to me
> >> that there is no use to carry DTMF OOB.
> >>
> >> SIP authors, can you answer why SIP INFO for carrying simple
> >> DTMF is not a standard solution ?
> >>
> >> Thanks,
> >> Rohan Kumar
> >>
> >>
> >> ----- Original Message -----
> >> From: "Attila Sipos" <AttilaS@vegastream.com>
> >> To: <rizsanyi@neobee.net>; <sip-implementors@cs.columbia.edu>
> >> Sent: Friday, May 09, 2003 9:50 AM
> >> Subject: RE: [Sip-implementors] dtmf digits using INFO
> >>
> >>
> >>>
> >>> Hello Zsolt,
> >>>
> >>> I still don't think there is a fully-agreed
> >>> standard. However, as much as I hate to say it,
> >>> CISCO have their own proposed standard which some
> >>> people (like us) interoperate with:
> >>>
> >>>
> >> http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/
> >> 122newft/122
> >>> t/122t11/ftinfo.htm
> >>>
> >>> Regards,
> >>>
> >>> Attila
> >>>
> >>> Attila Sipos
> >>> Software Engineer
> >>> <http://www.vegastream.com>
> >>> VegaStream : A World of difference for your Integrated=20
> Communications
> >>>
> >>> (jo napot kiv=E1nok)
> >>>
> >>>
> >>>
> >>>> -----Original Message-----
> >>>> From: Rizsanyi Zsolt [mailto:rizsanyi@myrealbox.com]
> >>>> Sent: 09 May 2003 13:41
> >>>> To: sip-implementors@cs.columbia.edu
> >>>> Subject: [Sip-implementors] dtmf digits using INFO
> >>>>
> >>>>
> >>>> Hi!
> >>>>
> >>>> Could somebody refer me to a current standard or draft, on
> >>>> how to send dtmf
> >>>> digits using the INFO method?
> >>>>
> >>>> Thanks
> >>>> Zsolt Rizsanyi
> >>>>
> >>>> _______________________________________________
> >>>> Sip-implementors mailing list
> >>>> Sip-implementors@cs.columbia.edu
> >>>> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
> >>>>
> >>>
> >>> _______________________________________________
> >>> Sip-implementors mailing list
> >>> Sip-implementors@cs.columbia.edu
> >>> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
> >>>
> >> _______________________________________________
> >> Sipping mailing list =20
> https://www1.ietf.org/mailman/listinfo/sipp> ing
> >> This list=20
> is for NEW development of the application=20
> of SIP
> >> Use sip-implementors@cs.columbia.edu for questions on current sip
> >> Use sip@ietf.org for new developments of core SIP
> >
> >
> > _______________________________________________
> > Sip-implementors mailing list
> > Sip-implementors@cs.columbia.edu
> > =20
> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors


_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

------_=3D_NextPart_001_01C3197C.55FF4E20
[Enclosed text/html ]
-----Original Message-----</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt; From: Rohan Mahy [ HREF=3D3D"mailto:rohan@cisco.com">mailto:rohan@cisco.com]</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt; Sent: Tuesday, May 13, 2003 1:34 =
AM</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt; To: SIP Developer</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt; Cc: sipping@ietf.org; Attila Sipos; =3D
</FONT>
<BR><FONT SIZE=3D3D2>&gt; sip-implementors@cs.columbia.edu;&nbsp; =
</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt; rizsanyi@neobee.net</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt; Subject: Re: [Sipping] Re: =3D
[Sip-implementors] dtmf digits </FONT>
<BR><FONT SIZE=3D3D2>&gt; using INFO&nbsp; </FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt; ... SIPauthors ... Can you answer please =
=3D
?</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
Hi Rohan,</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
The main reason to not carry DTMF in SIP INFOs is </FONT>
<BR><FONT SIZE=3D3D2>&gt; that there is&nbsp; </FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt; a very</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
good chance that you will get double-digits or otherwise&nbsp; </FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt; corrupted</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
digits because the signaling is not </FONT>
<BR><FONT SIZE=3D3D2>&gt; time-synchronized with the&nbsp; </FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt; audio</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
media.&nbsp; This issue is discussed in an expired draft which I&nbsp; =
=3D
</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt; wrote a</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
long time ago, which I abandoned in favor of the </FONT>
<BR><FONT SIZE=3D3D2>&gt; work generated&nbsp; </FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt; by the</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
app-interaction design team. At this point my draft </FONT>
<BR><FONT SIZE=3D3D2>&gt; is useful&nbsp; </FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt; only in</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
that it explains how double digits tend to occur.</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
my expired draft:</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; =3D
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt; HREF=3D3D"http://www.softarmor.com/sipping/drafts/morgue/draft-mahy-sipp=
in=3D
g-" =3D
TARGET=3D3D"_blank">http://www.softarmor.com/sipping/drafts/morgue/draft=
-m=3D
ahy-sipping-</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
signaled-digits-01.html</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
the app-interaction design team drafts:</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; =3D
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt; HREF=3D3D"http://www.softarmor.com/sipping/drafts/draft-burger-sipping-k=
pm=3D
l-" =3D
TARGET=3D3D"_blank">http://www.softarmor.com/sipping/drafts/draft-burger=
-s=3D
ipping-kpml- </FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt; 01.txt</FONT>
<BR><FONT SIZE=3D3D2>&gt; =3D
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt; HREF=3D3D"http://www.softarmor.com/sipping/drafts/draft-culpepper-sippin=
g-=3D
app-" =3D
TARGET=3D3D"_blank">http://www.softarmor.com/sipping/drafts/draft-culpep=
pe=3D
r-sipping-app-</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
interact-reqs-03.txt</FONT>
<BR><FONT SIZE=3D3D2>&gt; =3D
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt; HREF=3D3D"http://www.softarmor.com/sipping/drafts/draft-jennings-sip-app=
-i=3D
nfo-" =3D
TARGET=3D3D"_blank">http://www.softarmor.com/sipping/drafts/draft-jennin=
gs=3D
-sip-app-info-</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
00.txt</FONT>
<BR><FONT SIZE=3D3D2>&gt; =3D
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt; HREF=3D3D"http://www.softarmor.com/sipping/drafts/draft-rosenberg-sippin=
g-=3D
app-" =3D
TARGET=3D3D"_blank">http://www.softarmor.com/sipping/drafts/draft-rosenb=
er=3D
g-sipping-app-</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
interaction-framework-00.txt</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
thanks,</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
-rohan (Mahy)</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
On Friday, May 9, 2003, at 11:05 AM, SIP Developer wrote:</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; Hi All,</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; I hate to say that but SIP community =
=3D
doesn't agree</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; or think that there are lots of use =
=3D
and requirement for carrying</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; DTMF Out-of-Band.</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; Every month, I can see that someone =
=3D
asks the same question </FONT>
<BR><FONT SIZE=3D3D2>&gt; again and</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; again</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; that how do we send DTMF via INFO =
and =3D
Out-Of-Band .</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; 3pcc needs it, lots of service =3D
creation environment needs it and</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; doesn't it looks promising to have a =
=3D
way in SIP to do OOB</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; DTMF just like h245-alphanumeric in =
=3D
H.323.</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; If we think of applications,&nbsp; =
=3D
then there are lots of them </FONT>
<BR><FONT SIZE=3D3D2>&gt; that I can</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; count which doesn't care about the =
=3D
timing info carried by RFC2833.</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; And if we really care about timing, =
=3D
then carrying RFC2833 via</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; notify (Again OOB) is =3D
promising.</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; I really do not understand if this =
is =3D
a political issue to </FONT>
<BR><FONT SIZE=3D3D2>&gt; not promote</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; SIP INFO for carrying DTMF or if =3D
someone can justify it to me</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; that there is no use to carry DTMF =
=3D
OOB.</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; SIP authors, can you answer why SIP =
=3D
INFO for carrying simple</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; DTMF is not a standard solution =3D
?</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; Thanks,</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; Rohan Kumar</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; ----- Original Message -----</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; From: &quot;Attila Sipos&quot; =3D
&lt;AttilaS@vegastream.com&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; To: &lt;rizsanyi@neobee.net&gt;; =3D
&lt;sip-implementors@cs.columbia.edu&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; Sent: Friday, May 09, 2003 9:50 =3D
AM</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; Subject: RE: [Sip-implementors] dtmf =
=3D
digits using INFO</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt; Hello Zsolt,</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt; I still don't think there is a =
=3D
fully-agreed</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt; standard.&nbsp; However, as much =
=3D
as I hate to say it,</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt; CISCO have their own proposed =
=3D
standard which some</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt; people (like us) interoperate =
=3D
with:</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; HREF=3D3D"http://www.cisco.com/univercd/cc/td/doc/product/software/ios12=
2/=3D
" =3D
TARGET=3D3D"_blank">http://www.cisco.com/univercd/cc/td/doc/product/soft=
wa=3D
re/ios122/</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; 122newft/122</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt; t/122t11/ftinfo.htm</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt; Regards,</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt; Attila</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt; Attila Sipos</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt; Software Engineer</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt; &lt; HREF=3D3D"http://www.vegastream.com" =3D
TARGET=3D3D"_blank">http://www.vegastream.com&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt; VegaStream : A World of =
difference =3D
for your Integrated </FONT>
<BR><FONT SIZE=3D3D2>&gt; Communications</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt; (jo napot kiv=3DE1nok)</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt;&gt; -----Original =3D
Message-----</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt;&gt; From: Rizsanyi Zsolt [ HREF=3D3D"mailto:rizsanyi@myrealbox.com">mailto:rizsanyi@myrealbox.com A>=3D
]

> >>>> Sent: 09 May 2003 =
13:41

> >>>> To: =3D
sip-implementors@cs.columbia.edu

> >>>> Subject: [Sip-implementors] =
=3D
dtmf digits using INFO

> >>>>

> >>>>

> >>>> Hi!

> >>>>

> >>>> Could somebody refer me to a =
=3D
current standard or draft, on

> >>>> how to send dtmf

> >>>> digits using the INFO =3D
method?

> >>>>

> >>>> Thanks

> >>>> Zsolt Rizsanyi

> >>>>

> >>>> =3D
_______________________________________________

> >>>> Sip-implementors mailing =3D
list

> >>>> =3D
Sip-implementors@cs.columbia.edu

> >>>> HREF=3D3D"http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors=
" =3D
TARGET=3D3D"_blank">http://lists.cs.columbia.edu/mailman/listinfo/sip-im=
pl=3D
ementors</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt; =3D
_______________________________________________</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt; Sip-implementors mailing =3D
list</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt; =3D
Sip-implementors@cs.columbia.edu</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt; HREF=3D3D"http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors=
" =3D
TARGET=3D3D"_blank">http://lists.cs.columbia.edu/mailman/listinfo/sip-im=
pl=3D
ementors</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt;&gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; =3D
_______________________________________________</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; Sipping mailing list&nbsp; </FONT>
<BR><FONT SIZE=3D3D2>&gt; HREF=3D3D"https://www1.ietf.org/mailman/listinf=3D
o/sipp" =3D
TARGET=3D3D"_blank">https://www1.ietf.org/mailman/listinfo/sipp&gt; =
=3D
ing</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; This list </FONT>
<BR><FONT SIZE=3D3D2>&gt; is for NEW development of the application =3D
</FONT>
<BR><FONT SIZE=3D3D2>&gt; of SIP</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; Use sip-implementors@cs.columbia.edu =
=3D
for questions on current sip</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;&gt; Use sip@ietf.org for new =
developments =3D
of core SIP</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
_______________________________________________</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
Sip-implementors mailing list</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
Sip-implementors@cs.columbia.edu</FONT>
<BR><FONT SIZE=3D3D2>&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D
</FONT>
<BR><FONT SIZE=3D3D2>&gt; HREF=3D3D"http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors=
" =3D
TARGET=3D3D"_blank">http://lists.cs.columbia.edu/mailman/listinfo/sip-im=
pl=3D
ementors</FONT>
</P>
<BR>

<P><FONT =3D
SIZE=3D3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D3D2>Sipping mailing list&nbsp; HREF=3D3D"https://www1.ietf.org/mailman/listinfo/sipping" =3D
TARGET=3D3D"_blank">https://www1.ietf.org/mailman/listinfo/sipping</=
FO=3D
NT>
<BR><FONT SIZE=3D3D2>This list is for NEW development of the =
application =3D
of SIP</FONT>
<BR><FONT SIZE=3D3D2>Use sip-implementors@cs.columbia.edu for questions =
=3D
on current sip</FONT>
<BR><FONT SIZE=3D3D2>Use sip@ietf.org for new developments of core =3D
SIP</FONT>
</P>

</BODY>
</HTML>
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End of Sipping Digest
_______________________________________________
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16-May-2003 04:25:56 -0500,8877;000000000000-00000b1b
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From: "Shane McCormack" <smccormack@tssg.org>
To: "Sipping@Ietf. Org" <sipping@ietf.org>, "Sip@Ietf. Org" =
<sip@ietf.org>
Date: Tue, 13 May 2003 17:09:48 +0100
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Subject: [Sipping] PIDF format for spatial location
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This is a multi-part message in MIME format.

------=3D_NextPart_000_001F_01C31972.75B44360
Hi,

I have mailed the list before on this subject but I am still looking =
for a
bit more clarification on this subject. I need to retreive spatial =
location
in pidf format for work I am doing on my thesis.
I know that work is being done in the Geopriv working group on a format =
for
sending such information. Just as a matter of doing things correctly =
could
somebody with any knowledge help me in
by telling me the additional headers necessary, and any other thoughts =
so i
can put the correct information in my implementation and my thesis.

Thank you

Shane McCormack

Research Assistant,
TSSG, Confederation House,
Waterford Business Park, Cork Road,
Waterford, Co. Waterford.

Phone: +353 51 302927 Fax: +353 51 302901
Mobile: +353 87 9955505
http://www.tssg.org
MSN: mccormackshane@hotmail.com





------=3D_NextPart_000_001F_01C31972.75B44360
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_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

16-May-2003 04:26:56 -0500,3564;000000000000-00000b1c
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From: Nicole Johnson <NJ2@dataconnection.com>
To: "'sipping@ietf.org'" <sipping@ietf.org>
Date: Thu, 15 May 2003 15:23:23 +0100
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Subject: [Sipping] [Sipping]Questions on =
draft-ietf-sipping-connect-reuse-reqs-00
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I have a couple of questions about =
draft-ietf-sipping-connect-reuse-reqs-00.

1. Intended scope of alias parameter. =20

Is the aim of this draft to enable connections to be reused within a =
single
dialog, or across multiple dialogs, i.e. can an alias learned within =
the
scope of one dialog be used in another? Is the alias parameter valid =
on
out-of-dialog requests?

2. Target address fails to authenticate.

If a server receives a request containing the alias parameter and the =
target
address of the request fails authentication, how should the server =
react?
It could drop the message, respond to the target address with a "not
authorized" response, or continue processing the request as if the =
"alias"
parameter was not present. Is there a recommended behaviour?=20

Thanks,
Nicole


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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
This draft is a work item of the Session Initiation Proposal =
Investigation Working Group of the IETF.

                Title : SIP Generic Request History Capability =FB Requirements
                Author(s) : M. Barnes et al.
                Filename : draft-ietf-sipping-req-history-03.txt
                Pages : 12
                Date : 2003-5-15
=09
Many services that SIP is anticipated to support require the ability=20
to determine why and how the call arrived at a specific application. =20
Examples of such services include (but are not limited to) sessions=20
initiated to call centers via 'click to talk' SIP URLs on a web page,=20
'call history/logging' style services within intelligent 'call=20
management' software for SIP UAs and calls to voicemail servers and=20
call centers. While SIP implicitly provides the redirect/retarget=20
capabilities that enable calls to be routed to chosen applications,=20
there is currently no standard mechanism within SIP for communicating=20
the history of such a request. This 'request history' information=20
allows the receiving application to determine hints about how and why=20
the call arrived at the application/user. =20
This draft discusses the motivations in support of a mechanism for=20
recording the 'request history', and proposes detailed requirements=20
for such a generic 'request history' capability.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipping-req-history-03.tx=
t

To remove yourself from the IETF Announcement list, send a message to=20
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A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html=20
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
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In the body type:
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=09
NOTE: The mail server at ietf.org can return the document in
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                "multipart" MIME messages (i.e. documents which have been split
                up into multiple messages), so check your local documentation on
                how to manipulate these messages.
                =09
                =09
Below is the data which will enable a MIME compliant mail reader
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_______________________________________________
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Use sip@ietf.org for new developments of core SIP

16-May-2003 08:38:45 -0500,3724;000000000000-00000b1e
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I would like to take some issue with the voicemail example 10.2:

For this to work in any meaningful way, UA-A and UA-B must both use=20
UA-VM as their voicemail server. While this may be a reasonable=20
assumption in cases where both get phone service from the same =
provider,=20
it certainly doesn't work in all useful cases. For instance it doesn't=20
work if a call is forwarded from a business phone to a home phone.

I think at a minimum the limitations of this approach should be=20
mentioned. (Better would be to simply decide this is a poor application =

of History and remove the example.)

                Paul

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From: "Mary Barnes" <mbarnes@nortelnetworks.com>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>, sipping@ietf.org
Subject: RE: [Sipping] I-D ACTION:draft-ietf-sipping-req-history-03.txt
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Paul,

Your point is well taken. There is a statement upfront around the
limitations of the examples in the appendix, but I should probably
re-iterate that in section 10 and explicitly highlight the limitations =
of
the scenario in 10.2 as to under which situations it might work like =
this.
I would like to keep this example, as I do think it is illustrative of =
where
one could use the request history capability. It wasn't at all the
intention to be a prescriptive description of Voicemail services (which =
is
some of the wording I have in the last section of paragraph 1, which I =
will
re-emphasize in the appendix). Although, for the forwarding to home, I =
would
still think that it's a local policy decision at the proxy as to which =
VM it
would go to and that if you had forwarded to home that it would still =
go to
your office VM, which I think is consistent with the wording under the
example. However, I can see that some of the wording likely ought to be
changed as it is being too prescriptive around the service (and perhaps =
your
primary point), so I will also reword the text under the example flow. =
=20

Thanks for your input,
Mary. =20

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
Sent: Friday, May 16, 2003 8:19 AM
To: sipping@ietf.org
Subject: Re: [Sipping] I-D ACTION:draft-ietf-sipping-req-history-03.txt


I would like to take some issue with the voicemail example 10.2:

For this to work in any meaningful way, UA-A and UA-B must both use=20
UA-VM as their voicemail server. While this may be a reasonable=20
assumption in cases where both get phone service from the same =
provider,=20
it certainly doesn't work in all useful cases. For instance it doesn't=20
work if a call is forwarded from a business phone to a home phone.

I think at a minimum the limitations of this approach should be=20
mentioned. (Better would be to simply decide this is a poor application =

of History and remove the example.)

                Paul

_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP
_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

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Mary Barnes wrote:
>=20
> Although, for the forwarding to home, I would
> still think that it's a local policy decision at the proxy as to =
which VM it
> would go to and that if you had forwarded to home that it would still =
go to
> your office VM, which I think is consistent with the wording under =
the
> example.=20

Just to make sure we are making exactly the same point:

the example has the forwardee sending the request to the particular=20
voicemail server it deals with. Then that server is using the history=20
info to decide which of its mailboxes to use. In the office to home=20
case, home and office phones probably use different voicemail servers.=20
The voicemail server used at home will likely not contain a mailbox for =

the office address.

History just doesn't enter into a solution to this problem. Instead, =
the=20
home phone must not go to voicemail. It must fail back to the proxy for =

the office phone, which can then choose its own voicemail server and=20
mailbox.

                Paul

_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

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From: "Mary Barnes" <mbarnes@nortelnetworks.com>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>
Cc: sipping@ietf.org
Subject: Example 10.2 - should we remove it? (was RE: [Sipping] I-D =
ACTIO
                N:draft-ietf-sipping-req-history-03.txt
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Paul,

I agree for the scenario whereby there is a VM server at home, you =
likely
wouldn't need History-Info. This example, with the current =
introductory
wording, is clearly restricted to a single VM server in an enterprise
environment. So, perhaps your original suggestion to remove the =
example
altogether is well-taken. =20

Are there any other views? Everytime we've discussed this particular
example, the discussion has ratholed, so I don't want to digress into =
that,
but rather would like feedback on whether anyone has any problems with
removing this particular example? (We do have more flows in the =
solutions
document, as well, making them not nearly as useful as they were
originally). =20

Mary.=20

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
Sent: Friday, May 16, 2003 12:20 PM
To: Barnes, Mary [NGC:B602:EXCH]
Cc: sipping@ietf.org
Subject: Re: [Sipping] I-D ACTION:draft-ietf-sipping-req-history-03.txt




Mary Barnes wrote:
>=20
> Although, for the forwarding to home, I would
> still think that it's a local policy decision at the proxy as to =
which VM
it
> would go to and that if you had forwarded to home that it would still =
go
to
> your office VM, which I think is consistent with the wording under =
the
> example.=20

Just to make sure we are making exactly the same point:

the example has the forwardee sending the request to the particular=20
voicemail server it deals with. Then that server is using the history=20
info to decide which of its mailboxes to use. In the office to home=20
case, home and office phones probably use different voicemail servers.=20
The voicemail server used at home will likely not contain a mailbox for =

the office address.

History just doesn't enter into a solution to this problem. Instead, =
the=20
home phone must not go to voicemail. It must fail back to the proxy for =

the office phone, which can then choose its own voicemail server and=20
mailbox.

                Paul

_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

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From: "Rajarshi Chakraborty" <rajarshi_chakraborty@hotmail.com>
To: drage@lucent.com, ashishn@mahindrabt.com, =
asanth01@sprintspectrum.com,
      sip@ietf.org
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Subject: [Sipping] SIP mobility in WLAN
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Hi,

Where can I get call-flows for SIP session management in a wireless LAN =

(802.11)? 3GPP only addresses the issues for cellular networks. Or are =
they=20
too trivial to be addressed specially? I am specially interested in=20
situations of handoffs (in this case, the IP changes as soon as the =
host=20
shifts from the zone around one hotspot to another). Should SIP bother =
at=20
all to detect IP changes or should the control be completely outside =
SIP? I=20
mean should it be some other module that has the responsibility of =
informing=20
the SIP stack in the mobile host about the IP changes, so that the host =
may=20
then send a re-INVITE with a new SDP but the call still remains active?

regards
Rajarshi


Rajarshi Chakraborty
B-196, IIT Campus,
Kharagpur,
West Bengal, India
PIN-721302





----Original Message Follows----
From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Ashish Naik'" <ashishn@mahindrabt.com>, "Santharam, Arun [CC]"=20
<asanth01@sprintspectrum.com>, "Sip@Ietf. Org" <sip@ietf.org>
Subject: RE: [Sip] Session Mobility in SIP
Date: Thu, 8 May 2003 14:42:24 +0100

Your best starting point to see the architecture is 3GPP TS 23.228, =
with=20
reference to 3GPP TS 23.002 to see how these entities fit together into =
the=20
entire 3GPP architecture.

The SIP protocol application is covered in 3GPP TS 24.229 with example =
flows=20
in 3GPP TS 24.228.

These documents may be found at:

ftp://ftp.3gpp.org/Specs/latest/Rel-5/=20
<ftp://ftp.3gpp.org/Specs/latest/Rel-5/>

3GPP uses these specifications over GPRS to provide mobility.

There is a parallel set of 3GPP2 specifications, substantially of =
common=20
text, that uses the same architecture over mobile IP, instead of GPRS. =
These=20
documents are currently in 3GPP2 comment and review.

regards

Keith




Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com

-----Original Message-----
From: Ashish Naik [mailto:ashishn@mahindrabt.com]
Sent: 08 May 2003 06:23
To: Santharam, Arun [CC]; Sip@Ietf. Org
Subject: RE: [Sip] Session Mobility in SIP


ok. Is there any specific document on IETF and 3GPP that can explain =
how it=20
is being addressed ?


Ragards,
Ashish Naik
CoE Mobile Computing
-----------------------------------
Phone 4018100 Ext 3070


-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of =
Santharam,=20
Arun [CC]
Sent: Wednesday, April 09, 2003 9:42 PM
To: Ashish Naik; Sip@Ietf. Org
Subject: RE: [Sip] Session Mobility in SIP



Ashish,



Please find comments inline.



Thank you

arun



-----Original Message-----
From: Ashish Naik [mailto:ashishn@mahindrabt.com]
Sent: Wednesday, April 09, 2003 4:01 AM
To: Sip@Ietf. Org
Subject: [Sip] Session Mobility in SIP



How is SIP going to address session mobility ? Is there any initiative =
taken=20
up for this ?



  > This has been addressed in 3GPP standards. Please refer to the 3GPP =
All=20
IP architecture specifications.



SIP is said to be a better alternative than Mobile IP. Can I have some=20
latest development document references ?





  > SIP is not going to replace Mobile IP. It is going to leverage =
Mobile IP=20
for session management in 3G wireless networks.





Regards,
Ashish Naik
-------------------
CoE Mobile Computing
Phone: 4018100 Ext 3070




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19-May-2003 02:41:28 -0500,2687;000000000001-00000b23
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Date: Mon, 19 May 2003 03:40:17 -0400
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To: Juha Heinanen <jh@lohi.eng.song.fi>
CC: "Peterson, Jon" <jon.peterson@neustar.biz>,
      "'Conroy, Lawrence (SMTP)'" <lwc@roke.co.uk>, Rohan Mahy =
<rohan@cisco.com>,
      "'sipping@ietf.org'" <sipping@ietf.org>, bcampbell@dynamicsoft.com,
      "'Dean Willis'" <dean.willis@softarmor.com>, =
gonzalo.Camarillo@ericsson.com
Subject: Re: [Sipping] WGLC for draft-ietf-sipping-e164-02.txt
References: =
<0449D80A0E9B614A83FA9031B07E8D3B257A78@stntexch2.va.neustar.com> =
<16067.37072.865297.570579@harjus.eng.song.fi>
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Juha Heinanen wrote:
> Peterson, Jon writes:
>=20
> > I don't
> > think it is too much to ask that SIP applications utilizing ENUM =
be generous
> > in what they receive with regard to enumservices - it just =
doesn't seem
> > costly to be backwards compatible with the original iteration of =
the ENUM
> > standard.
>=20
> check, for example, with cisco and let me know when they accept enum
> replies with both formats. today they don't which is a very big
> problem.=20

Sounds like a good reason to document that you should be supporting=20
both. I think recommending that has only benefits and no drawbacks.

-Jonathan R.

--=20
Jonathan D. Rosenberg, Ph.D. 600 Lanidex Plaza
Chief Scientist Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com FAX: (973) 952-5050
http://www.jdrosen.net PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Subject: Re: [Sipping] WGLC for draft-ietf-sipping-e164-02.txt
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Juha Heinanen wrote:
> Peterson, Jon writes:
>=20
> > I don't
> > think it is too much to ask that SIP applications utilizing ENUM =
be generous
> > in what they receive with regard to enumservices - it just =
doesn't seem
> > costly to be backwards compatible with the original iteration of =
the ENUM
> > standard.
>=20
> check, for example, with cisco and let me know when they accept enum
> replies with both formats. today they don't which is a very big
> problem.=20

Sounds like a good reason to document that you should be supporting=20
both. I think recommending that has only benefits and no drawbacks.

-Jonathan R.

--=20
Jonathan D. Rosenberg, Ph.D. 600 Lanidex Plaza
Chief Scientist Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com FAX: (973) 952-5050
http://www.jdrosen.net PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

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      "'Conroy, Lawrence (SMTP)'" <lwc@roke.co.uk>, Rohan Mahy =
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      "'Dean Willis'" <dean.willis@softarmor.com>, =
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Subject: Re: [Sipping] WGLC for draft-ietf-sipping-e164-02.txt
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Jonathan Rosenberg writes:

  > Sounds like a good reason to document that you should be supporting=20
  > both. I think recommending that has only benefits and no drawbacks.

yes, we have way passed the point where the "old" service field format
need not be supported. because the "old" format must be supported for
ever, the question raises: why does it make sense to define the new
format at all.

-- juha

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      "'Conroy, Lawrence (SMTP)'" <lwc@roke.co.uk>, Rohan Mahy =
<rohan@cisco.com>,
      "'sipping@ietf.org'" <sipping@ietf.org>, bcampbell@dynamicsoft.com,
      "'Dean Willis'" <dean.willis@softarmor.com>, =
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Subject: Re: [Sipping] WGLC for draft-ietf-sipping-e164-02.txt
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<0449D80A0E9B614A83FA9031B07E8D3B257A78@stntexch2.va.neustar.com>
                <16067.37072.865297.570579@harjus.eng.song.fi>
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Jonathan Rosenberg writes:

  > Sounds like a good reason to document that you should be supporting=20
  > both. I think recommending that has only benefits and no drawbacks.

yes, we have way passed the point where the "old" service field format
need not be supported. because the "old" format must be supported for
ever, the question raises: why does it make sense to define the new
format at all.

-- juha
_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

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Date: Mon, 19 May 2003 04:14:55 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: Re: [Sipping] conf: comments on =
draft-rosenberg-sipping-conferenc
    ing-framework-01
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response inline.

Marjou Xavier wrote:
>>> Something is not clear to me : is CPCP really mandatory to
>>> perform something as "simple" as calling many recipients via a
>>> focus (adhoc conference).
>=20
>=20
>> In an ad-hoc conference, the focus resides with one of the
>> endpoints. Therefore, an internal interface can be used to
>> instruct the focus to call out to additional users. So, in the
>> case of ad-hoc, something as "simple" as mass invitation does not
>> require CPCP.
>=20
>=20
>>> With a centralized conference server, a client can use a series
>>> of REFER requests to bring new users into the conference. This
>>> is described in the document as a SIP mecahnism for adding
>>> users. However, for a large number of invitations, this is
>>> really inefficient, and thus a real control protocol like CPCP
>>> is needed.
>=20
>=20
> Is a CPCP body inside a SIP INVITE acceptable ? (I would like to
> avoid an additional round-trip delay for a CPCP request before the
> SIP request)

What additional round trip delay? If you use CPCP it would mean you=20
dont use refer.

-Jonathan R.

--=20
Jonathan D. Rosenberg, Ph.D. 600 Lanidex Plaza
Chief Scientist Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com FAX: (973) 952-5050
http://www.jdrosen.net PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

19-May-2003 03:39:50 -0500,6462;000000000001-00000b28
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Date: Mon, 19 May 2003 04:29:42 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Eric Burger wrote:

>> Having a URI with a mnemonic user part does not mean its no
>> longer opaque to the system, no longer unique, or that the intent
>> is for it to be guessed. For long lived conferences in
>> particular, where there is a likelihood that someone will need to
>> convey one over a phone, I see value in them being mnemonic.
>=20
>=20
> I think we agree here. That is: 1. There is value to having a
> mnemonic URI 2. The URI is opaque to the protocol.
>=20
> I was just pointing out that the text of the draft explicitly
> states the conference URI must not be a mnemonic (see quote above).

I think the new text in sipping-00 is OK.

>=20
>=20
> [snip]
>=20
>=20
>>> Section 5.2.1 Is it worth mentioning a stateful application
>>=20
>> server as
>>=20
>>> a mechanism for running a conference?
>>=20
>> I don't understand. 5.2.1 is talking about adding users to a=20
>> conference. What does that have to do with using an application
>> server?
>=20
>=20
> All INVITEs come from the application server. This means the
> application server is fully aware of the state of all call legs;
> can directly manipulate state, permissions, and media policy; and
> has the added bonus of making CPCP entirely unnecessary.
>=20
> Given that the largest conference service bureaux in the world are
> on the order of a few tens of thousands of ports, I don't think the
> "application server doesn't scale like a proxy server" argument
> really holds.

I still dont see what this has to do with section 5.2.1.

>=20
> [snip]
>=20
>=20
>>> Section 5.4.2 and 5.4.3 I don't really get the difference
>>> between these two examples.
>>=20
>> One of them uses CPCP to remove a user. This can be done by an
>> automata. The other uses a web application to remove a user. THis
>> can only be done by a user.
>=20
> [snip]
>=20
>> So, just because both SIP and CPCP can change the state of a=20
>> conference, does not mean that they are the same protocol.
>=20
> [snip]
>=20
>> CPCP, being a data transaction protocol, would also allow you to
>> directly query for the list. Just because there is this sliver
>> of overlap does not mean that they are the same protocol.
>=20
>=20
> A "sliver of overlap" is OK. A full set of overlap is not. Look
> at the draft, for just about every operation, the draft says, "Here
> is the way to do it with SIP, and here is the way to do it with
> CPCP." It may be my misreading, but it sounds like there is a lot
> of overlap.

Thats because the framework is not touching on the breadth of CPCP,=20
which would also set general conference policy (like who is allowed to=20
join, who can moderate, name of the conference, etc.). Those=20
conference operations which are of the flavor "create this dialog" can=20
be done with SIP, but only those operations.

-Jonathan R.


--=20
Jonathan D. Rosenberg, Ph.D. 600 Lanidex Plaza
Chief Scientist Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com FAX: (973) 952-5050
http://www.jdrosen.net PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

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International SIP 2004, January 20-23, Paris France

After a setback in 2002 over 2001, explained by the difficult economic =
context plaguing the telecom industry, International SIP 2003 clearly =
recaptured the interest of the VoIP community.=20
Don't miss the 2004 edition.

The call for papers dead line has been extendeed to June 15.
Please get all details at:
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_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
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19-May-2003 16:16:30 -0500,4773;000000000001-00000b2a
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Date: Mon, 19 May 2003 17:15:28 -0400
From: Flemming Andreasen <fandreas@cisco.com>
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CC: "'sipping@ietf.org'" <sipping@ietf.org>,
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Subject: Re: [Sipping] Last Call for early media use cases
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<F8EFC4B4A8C016428BC1F589296D4FBFBC1083@esealnt630.al.sw.ericsson.se>
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I believe it is useful to have a flag that indicates the "early media =
is coming", which would suggest to the UAC that it probably should not =
be
generating local ringback. Motivation for such a flag is to avoid the =
potential race between generating local ringback and having in-band =
media
turning such local ringback off. It applies equally well to the =
intelligent SIP endpoint and the decomposed gateway, although it may be =
more obvious
for the decomposed gateway. You may argue, that signaling should be =
faster than media, that you can have a small delay in applying the =
local
ringback, etc, however they all amount to heuristics. Also, consider =
that the far-end side may be using silence suppression, and the =
potential for
the race to occur becomes greater. By having (and using) this flag, the =
race is avoided.


This brings me to another issue. So far, the ringback/early media/media =
discussion has mainly focused on having media received turning local
ringback off. A great deal of concern has been expressed with the =
robustness of this principle and I agree with this concern. It turns =
out we
already have an example that breaks this model, namely comedia which =
specifies that:

"Endpoints that specify direction:active MUST be prepared to receive on =
the ports from which they send. Once they learn the IP address and =
port of
their peer from the peer's SDP, they SHOULD immediately send some kind =
of media (even if just comfort noise) to each of these ports."

In other words, use of comedia may turn off local ringback. In general, =
NAT/firewall traversal incurs this problem as it may need to refresh
bindings/pinholes (as described in STUN) before a call has been =
answered, and there will probably be other cases too. This strongly =
suggests that
the current "local ringaback turned off by early media" design is =
simply too fragile. We could for example:
1. Have a way of determining/indicating that some early media packets =
should not override local ringback or whatever else might be going on =
(this
could potentially just be by convention, e.g. silence suppression =
packets should not affect local ringback, which would then be a =
sanctioned way of
doing RTP connectivity things without affecting early media)
2. Have a way of turning local ringback back on after it has been =
turned off due to receipt of early media (consider the case where =
forking lead to
multiple early dialogs, but just one early media stream, and all of a =
sudden that early media stream disappers).

It would be good if the early media draft could address these issues as =
well.

-- Flemming


"Gonzalo Camarillo Gonzalez (LMF)" wrote:

> Folks,
>
> in the last IETF meeting, we requested early media use cases that =
show the need of a "media coming" flag. So far, we have not received =
anything.
>
> We intend to release a new version of the early media draft in a week =
or so. So, that is the time you have to send me your use cases.
>
> Thanks,
>
> Gonzalo
> _______________________________________________
> Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP

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Date: Mon, 19 May 2003 17:15:28 -0400
From: Flemming Andreasen <fandreas@cisco.com>
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<Gonzalo.Camarillo@lmf.ericsson.se>
CC: "'sipping@ietf.org'" <sipping@ietf.org>,
      "Jon Peterson (E-mail)" <jon.peterson@neustar.biz>,
      "Rohan Mahy (E-mail)" <rohan@cisco.com>,
      "Dean Willis (E-mail)" <dean.willis@softarmor.com>
Subject: Re: [Sipping] Last Call for early media use cases
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I believe it is useful to have a flag that indicates the "early media =
is coming", which would suggest to the UAC that it probably should not =
be
generating local ringback. Motivation for such a flag is to avoid the =
potential race between generating local ringback and having in-band =
media
turning such local ringback off. It applies equally well to the =
intelligent SIP endpoint and the decomposed gateway, although it may be =
more obvious
for the decomposed gateway. You may argue, that signaling should be =
faster than media, that you can have a small delay in applying the =
local
ringback, etc, however they all amount to heuristics. Also, consider =
that the far-end side may be using silence suppression, and the =
potential for
the race to occur becomes greater. By having (and using) this flag, the =
race is avoided.


This brings me to another issue. So far, the ringback/early media/media =
discussion has mainly focused on having media received turning local
ringback off. A great deal of concern has been expressed with the =
robustness of this principle and I agree with this concern. It turns =
out we
already have an example that breaks this model, namely comedia which =
specifies that:

"Endpoints that specify direction:active MUST be prepared to receive on =
the ports from which they send. Once they learn the IP address and =
port of
their peer from the peer's SDP, they SHOULD immediately send some kind =
of media (even if just comfort noise) to each of these ports."

In other words, use of comedia may turn off local ringback. In general, =
NAT/firewall traversal incurs this problem as it may need to refresh
bindings/pinholes (as described in STUN) before a call has been =
answered, and there will probably be other cases too. This strongly =
suggests that
the current "local ringaback turned off by early media" design is =
simply too fragile. We could for example:
1. Have a way of determining/indicating that some early media packets =
should not override local ringback or whatever else might be going on =
(this
could potentially just be by convention, e.g. silence suppression =
packets should not affect local ringback, which would then be a =
sanctioned way of
doing RTP connectivity things without affecting early media)
2. Have a way of turning local ringback back on after it has been =
turned off due to receipt of early media (consider the case where =
forking lead to
multiple early dialogs, but just one early media stream, and all of a =
sudden that early media stream disappers).

It would be good if the early media draft could address these issues as =
well.

-- Flemming


"Gonzalo Camarillo Gonzalez (LMF)" wrote:

> Folks,
>
> in the last IETF meeting, we requested early media use cases that =
show the need of a "media coming" flag. So far, we have not received =
anything.
>
> We intend to release a new version of the early media draft in a week =
or so. So, that is the time you have to send me your use cases.
>
> Thanks,
>
> Gonzalo
> _______________________________________________
> Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP

_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

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From: "Ejzak, Richard P (Richard)" <ejzak@lucent.com>
To: "'Gonzalo Camarillo Gonzalez (LMF)'"
                  <Gonzalo.Camarillo@lmf.ericsson.se>,
      "'sipping@ietf.org'"
                  <sipping@ietf.org>
Cc: "Jon Peterson (E-mail)" <jon.peterson@neustar.biz>,
      "Rohan Mahy (E-mail)" <rohan@cisco.com>,
      "Dean Willis (E-mail)"
                  <dean.willis@softarmor.com>
Subject: RE: [Sipping] Last Call for early media use cases
Date: Mon, 19 May 2003 16:45:07 -0500
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------_=3D_NextPart_000_01C31E4F.EA121D72
Gonzalo,
Please excuse my tardiness in responding to your request for early =
media use cases. I believe the attachment includes an example =
configuration and call flow that clearly illustrates some problems that =
need to be solved with the existing SIP early media procedures.

Thanks,

Richard Ejzak
Lucent Technologies

-----Original Message-----
From: Gonzalo Camarillo Gonzalez (LMF)
[mailto:Gonzalo.Camarillo@lmf.ericsson.se]
Sent: Wednesday, May 14, 2003 2:13 AM
To: 'sipping@ietf.org'
Cc: Jon Peterson (E-mail); Rohan Mahy (E-mail); Dean Willis (E-mail)
Subject: [Sipping] Last Call for early media use cases


Folks,

in the last IETF meeting, we requested early media use cases that show =
the need of a "media coming" flag. So far, we have not received =
anything.

We intend to release a new version of the early media draft in a week =
or so. So, that is the time you have to send me your use cases.

Thanks,

Gonzalo
_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP


------_=3D_NextPart_000_01C31E4F.EA121D72
Early Media Use Case

The following figure is an example call flow highlighting a range of =
issues associated with control of early media. In this scenario, A and =
D are gateway controllers performing interworking with other networks =
using in-band call progress information, B is a network feature server =
performing two call forwarding procedures (C to D and then D to E), and =
C and E are SIP endpoints providing only out-of-band call progress =
information. B is required to behave as a B2BUA since gateways are =
known to have difficulties with SIP forking procedures, which would =
otherwise be required in this forwarding scenario for B to behave as a =
SIP proxy. C and E are shown performing delayed SDP negotiation, which =
may occur due to resource reservation delays at a mobile endpoint or =
for other reasons. The scenario also assumes the use of a SIP profile =
including the use of reliable provisional responses and preconditions, =
although PRACK transactions without SDP and other details are not =
shown. 181 responses may occur during the scenario to report =
forwarding, but are not shown since they do not affect the processing =
of early media. Cleanup of dialogs with C and D after forwarding using =
CANCELs and/or final responses is also not shown to limit clutter in =
the figure.


A B C D =
        E
|(1) INVITE sdp1 |(2) INVITE sdp1 | | =
        |
|----------------->|----------------->| | =
        |
|(4) 180 |(3) 180 | | =
        |
|<-----------------|<-----------------| | =
        |
| | | | =
        |
|(6) 183 sdp2 |(5) 183 sdp2 | | =
        |
|<-----------------|<-----------------| | =
        |
| | | | =
        |
| | |(7) INVITE(no SDP)| =
        |
| |------------------------------------>| =
        |
|(9) 183 | |(8) 183 sdp3 | =
        |
|<-----------------|<------------------------------------| =
        |
|(10) UPDATE sdp3 | | | =
        |
|<-----------------| | | =
        |
|(11) 200 OK sdp4 | |(12) PRACK sdp4 | =
        |
|----------------->|------------------------------------>| =
        |
| | |(13) 200 OK | =
        |
| |<------------------------------------| =
        |
| | | | =
        |
|(15) 180 | |(14) 180 | =
        |
|<-----------------|<------------------------------------| =
        |
| | | | =
        |
| | | |(16) =
INVITE(noSDP)|
| =
|------------------------------------------------------->|
|(18) 180 | | |(17) 180 =
        |
|<-----------------|<---------------------------------------------------=
----|
| | | | =
=3D
        |
|(20) 183 | | |(19) 183 sdp5 =
=3D
        |
|<-----------------|<---------------------------------------------------=
=3D
----|
|(21) UPDATE sdp5 | | | =
=3D
        |
|<-----------------| | | =
=3D
        |
|(22) 200 OK sdp6 | | |(23) PRACK =
=3D
sdp6 |
|----------------->|----------------------------------------------------=
=3D
--->|
| | | |(24) 200 OK =
=3D
        |
| =
|<---------------------------------------------------=3D
----|
| | | | =
=3D
        |
|(26) 200 OK (INV) | | |(25) 200 OK =
=3D
(INV) |
|<-----------------|<---------------------------------------------------=
=3D
----|
|(27) ACK | | |(28) ACK =
=3D
        |
|----------------->|----------------------------------------------------=
=3D
--->|


At message (4), gateway controller A begins playing local ringing to =
=3D
the originating party (not shown) to represent alerting at C (which =3D
provides only out-of-band call progress information). Some time later, =
=3D
when C is able to respond to the media offer in (2), it sends a 183 =3D
response with SDP (5) to B, which sends another 183 response with SDP =
=3D
(6) to A, completing SDP negotiation between A and C. This is the =3D
first case in which an early media issue arises. A must be prepared to =
=3D
override local ringing in the presence of in-band media to avoid the =
=3D
potential for media clipping at called party answer. To avoid =3D
premature termination of local ringing after (6), C must avoid sending =
=3D
any media until called party answer occurs. There should be a =3D
documented convention covering precisely what triggers the transition =
=3D
to in-band media - whether it be the existence of any RTP packets or =
=3D
the existence of media content in RTP packets above some noise =3D
threshold. The use of a noise threshold would allow RTP playout =3D
adjustments to occur prior to the transmission of any content, and is =
=3D
preferred.

At some point after (6), B performs forwarding from C to gateway =3D
controller D. D establishes new media at (8) and provides indication =
=3D
of called party alerting at (14). Following SDP offer/answer rules for =
=3D
reliable provisional responses in RFC 3262, RFC 3264 and RFC 3311, B =
=3D
separates sdp3 from (8) and sends the 183 and sdp3 to A in messages (9) =
=3D
and (10), respectively. This is the second case in which an early =3D
media issue arises. A has no basis to treat the media established with =
=3D
D in (10) and (11) any differently from the media established with C in =
=3D
(6). So A must rely on the presence of media (however this is defined) =
=3D
from D for A to stop playing local ringing and pass in-band call =3D
progress information from D to the calling party. If a noise threshold =
=3D
is used to identify the presence of in-band media and in-band call =3D
progress information from D begins with a protracted period of silence =
=3D
(noise), due to various possible network conditions, then A will =3D
continue to play local ringing even though silence would provide a more =
=3D
accurate indication of call progress.

When called party alerting indication comes from D at (14) and (15), A =
=3D
would presumably not change early media handling. Either in-band media =
=3D
was detected after (11) and local ringing was turned off in preference =
=3D
to in-band media, or no in-band media was detected after (11) and local =
=3D
ringing continued.

But in this scenario sometime after (15), B performs forwarding from D =
=3D
to endpoint E (which provides only out-of-band call progress =3D
information). The 180 from E (18) should immediately trigger the =3D
playing of local ringing at A, but A has no basis to distinguish the =
=3D
condition at (18) from the condition at (15). This is the third early =
=3D
media issue. A may be able to deduce from the absence of media from D =
=3D
that it should start generating local ringing at (18), but there =3D
appears to be no way to do this reliably. There are seconds of silence =
=3D
even during active in-band call progress information, so it's not clear =
=3D
what would trigger A to begin local ringing. The early media issues =
=3D
associated with the remainder of the scenario have already been =3D
discussed for messages (5) and (6).

Early Media Issues

Summarizing the three early media issues described in the preceding =3D
section:

1) A gateway playing local ringing in response to receipt of 180 should =
=3D
be prepared to give priority to media from the network at answer to =3D
avoid clipping. In-band media detection should be based on the =3D
presence of media content rather than just the presence of RTP packets =
=3D
to avoid waiting for RTP playout adjustments.

2) A similar technique to 1) could handle a transition from local =3D
ringing to in-band call progress information. Unfortunately, this =3D
technique is not robust in detecting this transition when the intended =
=3D
in-band information is silence. A gateway may replace periods of =3D
intended silence with local ringing.

3) A gateway passing in-band call progress information has difficulty =
=3D
identifying when it should transition to playing local ringing. Using =
=3D
the absence of in-band call progress information as a trigger to =3D
initiate local ringing is unreliable and subject to significant delay.

Potential Solution

Problems 2) and 3) in the preceding section can be readily solved with =
=3D
the introduction of one minor change: a new piece of information =3D
associated with any 18x response to indicate when in-band call progress =
=3D
information is available. The following figure uses the same scenario =
=3D
in the preceding use case but includes the use of the in-band (IB) =3D
indication in the 18x responses as appropriate. This solution is =3D
presented for illustrative purposes only. Other solutions also exist =
=3D
but all require the addition of some new information to one or more 18x =
=3D
responses.


A B C D =
=3D
        E
|(1) INVITE sdp1 |(2) INVITE sdp1 | | =
=3D
        |
|----------------->|----------------->| | =
=3D
        |
|(4) 180 |(3) 180 | | =
=3D
        |
|<-----------------|<-----------------| | =
=3D
        |
| | | | =
=3D
        |
|(6) 183 sdp2 |(5) 183 sdp2 | | =
=3D
        |
|<-----------------|<-----------------| | =
=3D
        |
| | | | =
=3D
        |
| | |(7) INVITE(no SDP)| =
=3D
        |
| |------------------------------------>| =
=3D
        |
|(9) 183 IB | |(8) 183 IB sdp3 | =
=3D
        |
|<-----------------|<------------------------------------| =
=3D
        |
|(10) UPDATE sdp3 | | | =
=3D
        |
|<-----------------| | | =
=3D
        |
|(11) 200 OK sdp4 | |(12) PRACK sdp4 | =
=3D
        |
|----------------->|------------------------------------>| =
=3D
        |
| | |(13) 200 OK | =
=3D
        |
| |<------------------------------------| =
=3D
        |
| | | | =
=3D
        |
|(15) 180 IB | |(14) 180 IB | =
=3D
        |
|<-----------------|<------------------------------------| =
=3D
        |
| | | | =
=3D
        |
| | | |(16) =3D
INVITE(noSDP)|
| =3D
|------------------------------------------------------->|
|(18) 180 | | |(17) 180 =
=3D
        |
|<-----------------|<---------------------------------------------------=
=3D
----|
| | | | =
=3D
        |
|(20) 183 | | |(19) 183 sdp5 =
=3D
        |
|<-----------------|<---------------------------------------------------=
=3D
----|
|(21) UPDATE sdp5 | | | =
=3D
        |
|<-----------------| | | =
=3D
        |
|(22) 200 OK sdp6 | | |(23) PRACK =
=3D
sdp6 |
|----------------->|----------------------------------------------------=
=3D
--->|
| | | |(24) 200 OK =
=3D
        |
| =3D
|<-------------------------------------------------------|
| | | | =
=3D
        |
|(26) 200 OK (INV) | | |(25) 200 OK =
=3D
(INV) |
|<-----------------|<---------------------------------------------------=
=3D
----|
|(27) ACK | | |(28) ACK =
=3D
        |
|----------------->|----------------------------------------------------=
=3D
--->|


A interprets 183 (6) as an indication that local ringing should =3D
continue until called party media is detected since there is no IB =3D
indication. 183 IB (8) and 183 IB (9) cause A to stop local ringing in =
=3D
preference to in-band media. There is no possibility of missing =3D
intended periods of in-band silence. 180 IB (14) and 180 IB (15) =3D
clearly indicate to A that there is called party alerting but the =3D
in-band information should be passed to the calling party rather than =
=3D
local ringing. Finally, 180 (18) and 183 (20) clearly indicate that A =
=3D
should generate local ringing until called party answer or some other =
=3D
change in early media handling. This flow is identical to the original =
=3D
flow except for the addition of the IB indication to four of the =3D
provisional responses in (8), (9), (14) and (15).
------_=3D_NextPart_000_01C31E4F.EA121D72--

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Message-ID: =
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From: "Ejzak, Richard P (Richard)" <ejzak@lucent.com>
To: "'Gonzalo Camarillo Gonzalez (LMF)'"
                  <Gonzalo.Camarillo@lmf.ericsson.se>,
      "'sipping@ietf.org'"
                  <sipping@ietf.org>
Cc: "Jon Peterson (E-mail)" <jon.peterson@neustar.biz>,
      "Rohan Mahy (E-mail)" <rohan@cisco.com>,
      "Dean Willis (E-mail)"
                  <dean.willis@softarmor.com>
Subject: RE: [Sipping] Last Call for early media use cases
Date: Mon, 19 May 2003 16:45:07 -0500
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------_=3D_NextPart_000_01C31E4F.EA121D72
Gonzalo,
Please excuse my tardiness in responding to your request for early =
media use cases. I believe the attachment includes an example =
configuration and call flow that clearly illustrates some problems that =
need to be solved with the existing SIP early media procedures.

Thanks,

Richard Ejzak
Lucent Technologies

-----Original Message-----
From: Gonzalo Camarillo Gonzalez (LMF)
[mailto:Gonzalo.Camarillo@lmf.ericsson.se]
Sent: Wednesday, May 14, 2003 2:13 AM
To: 'sipping@ietf.org'
Cc: Jon Peterson (E-mail); Rohan Mahy (E-mail); Dean Willis (E-mail)
Subject: [Sipping] Last Call for early media use cases


Folks,

in the last IETF meeting, we requested early media use cases that show =
the need of a "media coming" flag. So far, we have not received =
anything.

We intend to release a new version of the early media draft in a week =
or so. So, that is the time you have to send me your use cases.

Thanks,

Gonzalo
_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP


------_=3D_NextPart_000_01C31E4F.EA121D72
Early Media Use Case

The following figure is an example call flow highlighting a range of =
issues associated with control of early media. In this scenario, A and =
D are gateway controllers performing interworking with other networks =
using in-band call progress information, B is a network feature server =
performing two call forwarding procedures (C to D and then D to E), and =
C and E are SIP endpoints providing only out-of-band call progress =
information. B is required to behave as a B2BUA since gateways are =
known to have difficulties with SIP forking procedures, which would =
otherwise be required in this forwarding scenario for B to behave as a =
SIP proxy. C and E are shown performing delayed SDP negotiation, which =
may occur due to resource reservation delays at a mobile endpoint or =
for other reasons. The scenario also assumes the use of a SIP profile =
including the use of reliable provisional responses and preconditions, =
although PRACK transactions without SDP and other details are not =
shown. 181 responses may occur during the scenario to report =
forwarding, but are not shown since they do not affect the processing =
of early media. Cleanup of dialogs with C and D after forwarding using =
CANCELs and/or final responses is also not shown to limit clutter in =
the figure.


A B C D =
        E
|(1) INVITE sdp1 |(2) INVITE sdp1 | | =
        |
|----------------->|----------------->| | =
        |
|(4) 180 |(3) 180 | | =
        |
|<-----------------|<-----------------| | =
        |
| | | | =
        |
|(6) 183 sdp2 |(5) 183 sdp2 | | =
        |
|<-----------------|<-----------------| | =
        |
| | | | =
        |
| | |(7) INVITE(no SDP)| =
        |
| |------------------------------------>| =
        |
|(9) 183 | |(8) 183 sdp3 | =
        |
|<-----------------|<------------------------------------| =
        |
|(10) UPDATE sdp3 | | | =
        |
|<-----------------| | | =
        |
|(11) 200 OK sdp4 | |(12) PRACK sdp4 | =
        |
|----------------->|------------------------------------>| =
        |
| | |(13) 200 OK | =
        |
| |<------------------------------------| =
        |
| | | | =
        |
|(15) 180 | |(14) 180 | =
        |
|<-----------------|<------------------------------------| =
        |
| | | | =
        |
| | | |(16) =
INVITE(noSDP)|
| =
|------------------------------------------------------->|
|(18) 180 | | |(17) 180 =
        |
|<-----------------|<---------------------------------------------------=
----|
| | | | =
=3D
        |
|(20) 183 | | |(19) 183 sdp5 =
=3D
        |
|<-----------------|<---------------------------------------------------=
=3D
----|
|(21) UPDATE sdp5 | | | =
=3D
        |
|<-----------------| | | =
=3D
        |
|(22) 200 OK sdp6 | | |(23) PRACK =
=3D
sdp6 |
|----------------->|----------------------------------------------------=
=3D
--->|
| | | |(24) 200 OK =
=3D
        |
| =
|<---------------------------------------------------=3D
----|
| | | | =
=3D
        |
|(26) 200 OK (INV) | | |(25) 200 OK =
=3D
(INV) |
|<-----------------|<---------------------------------------------------=
=3D
----|
|(27) ACK | | |(28) ACK =
=3D
        |
|----------------->|----------------------------------------------------=
=3D
--->|


At message (4), gateway controller A begins playing local ringing to =
=3D
the originating party (not shown) to represent alerting at C (which =3D
provides only out-of-band call progress information). Some time later, =
=3D
when C is able to respond to the media offer in (2), it sends a 183 =3D
response with SDP (5) to B, which sends another 183 response with SDP =
=3D
(6) to A, completing SDP negotiation between A and C. This is the =3D
first case in which an early media issue arises. A must be prepared to =
=3D
override local ringing in the presence of in-band media to avoid the =
=3D
potential for media clipping at called party answer. To avoid =3D
premature termination of local ringing after (6), C must avoid sending =
=3D
any media until called party answer occurs. There should be a =3D
documented convention covering precisely what triggers the transition =
=3D
to in-band media - whether it be the existence of any RTP packets or =
=3D
the existence of media content in RTP packets above some noise =3D
threshold. The use of a noise threshold would allow RTP playout =3D
adjustments to occur prior to the transmission of any content, and is =
=3D
preferred.

At some point after (6), B performs forwarding from C to gateway =3D
controller D. D establishes new media at (8) and provides indication =
=3D
of called party alerting at (14). Following SDP offer/answer rules for =
=3D
reliable provisional responses in RFC 3262, RFC 3264 and RFC 3311, B =
=3D
separates sdp3 from (8) and sends the 183 and sdp3 to A in messages (9) =
=3D
and (10), respectively. This is the second case in which an early =3D
media issue arises. A has no basis to treat the media established with =
=3D
D in (10) and (11) any differently from the media established with C in =
=3D
(6). So A must rely on the presence of media (however this is defined) =
=3D
from D for A to stop playing local ringing and pass in-band call =3D
progress information from D to the calling party. If a noise threshold =
=3D
is used to identify the presence of in-band media and in-band call =3D
progress information from D begins with a protracted period of silence =
=3D
(noise), due to various possible network conditions, then A will =3D
continue to play local ringing even though silence would provide a more =
=3D
accurate indication of call progress.

When called party alerting indication comes from D at (14) and (15), A =
=3D
would presumably not change early media handling. Either in-band media =
=3D
was detected after (11) and local ringing was turned off in preference =
=3D
to in-band media, or no in-band media was detected after (11) and local =
=3D
ringing continued.

But in this scenario sometime after (15), B performs forwarding from D =
=3D
to endpoint E (which provides only out-of-band call progress =3D
information). The 180 from E (18) should immediately trigger the =3D
playing of local ringing at A, but A has no basis to distinguish the =
=3D
condition at (18) from the condition at (15). This is the third early =
=3D
media issue. A may be able to deduce from the absence of media from D =
=3D
that it should start generating local ringing at (18), but there =3D
appears to be no way to do this reliably. There are seconds of silence =
=3D
even during active in-band call progress information, so it's not clear =
=3D
what would trigger A to begin local ringing. The early media issues =
=3D
associated with the remainder of the scenario have already been =3D
discussed for messages (5) and (6).

Early Media Issues

Summarizing the three early media issues described in the preceding =3D
section:

1) A gateway playing local ringing in response to receipt of 180 should =
=3D
be prepared to give priority to media from the network at answer to =3D
avoid clipping. In-band media detection should be based on the =3D
presence of media content rather than just the presence of RTP packets =
=3D
to avoid waiting for RTP playout adjustments.

2) A similar technique to 1) could handle a transition from local =3D
ringing to in-band call progress information. Unfortunately, this =3D
technique is not robust in detecting this transition when the intended =
=3D
in-band information is silence. A gateway may replace periods of =3D
intended silence with local ringing.

3) A gateway passing in-band call progress information has difficulty =
=3D
identifying when it should transition to playing local ringing. Using =
=3D
the absence of in-band call progress information as a trigger to =3D
initiate local ringing is unreliable and subject to significant delay.

Potential Solution

Problems 2) and 3) in the preceding section can be readily solved with =
=3D
the introduction of one minor change: a new piece of information =3D
associated with any 18x response to indicate when in-band call progress =
=3D
information is available. The following figure uses the same scenario =
=3D
in the preceding use case but includes the use of the in-band (IB) =3D
indication in the 18x responses as appropriate. This solution is =3D
presented for illustrative purposes only. Other solutions also exist =
=3D
but all require the addition of some new information to one or more 18x =
=3D
responses.


A B C D =
=3D
        E
|(1) INVITE sdp1 |(2) INVITE sdp1 | | =
=3D
        |
|----------------->|----------------->| | =
=3D
        |
|(4) 180 |(3) 180 | | =
=3D
        |
|<-----------------|<-----------------| | =
=3D
        |
| | | | =
=3D
        |
|(6) 183 sdp2 |(5) 183 sdp2 | | =
=3D
        |
|<-----------------|<-----------------| | =
=3D
        |
| | | | =
=3D
        |
| | |(7) INVITE(no SDP)| =
=3D
        |
| |------------------------------------>| =
=3D
        |
|(9) 183 IB | |(8) 183 IB sdp3 | =
=3D
        |
|<-----------------|<------------------------------------| =
=3D
        |
|(10) UPDATE sdp3 | | | =
=3D
        |
|<-----------------| | | =
=3D
        |
|(11) 200 OK sdp4 | |(12) PRACK sdp4 | =
=3D
        |
|----------------->|------------------------------------>| =
=3D
        |
| | |(13) 200 OK | =
=3D
        |
| |<------------------------------------| =
=3D
        |
| | | | =
=3D
        |
|(15) 180 IB | |(14) 180 IB | =
=3D
        |
|<-----------------|<------------------------------------| =
=3D
        |
| | | | =
=3D
        |
| | | |(16) =3D
INVITE(noSDP)|
| =3D
|------------------------------------------------------->|
|(18) 180 | | |(17) 180 =
=3D
        |
|<-----------------|<---------------------------------------------------=
=3D
----|
| | | | =
=3D
        |
|(20) 183 | | |(19) 183 sdp5 =
=3D
        |
|<-----------------|<---------------------------------------------------=
=3D
----|
|(21) UPDATE sdp5 | | | =
=3D
        |
|<-----------------| | | =
=3D
        |
|(22) 200 OK sdp6 | | |(23) PRACK =
=3D
sdp6 |
|----------------->|----------------------------------------------------=
=3D
--->|
| | | |(24) 200 OK =
=3D
        |
| =3D
|<-------------------------------------------------------|
| | | | =
=3D
        |
|(26) 200 OK (INV) | | |(25) 200 OK =
=3D
(INV) |
|<-----------------|<---------------------------------------------------=
=3D
----|
|(27) ACK | | |(28) ACK =
=3D
        |
|----------------->|----------------------------------------------------=
=3D
--->|


A interprets 183 (6) as an indication that local ringing should =3D
continue until called party media is detected since there is no IB =3D
indication. 183 IB (8) and 183 IB (9) cause A to stop local ringing in =
=3D
preference to in-band media. There is no possibility of missing =3D
intended periods of in-band silence. 180 IB (14) and 180 IB (15) =3D
clearly indicate to A that there is called party alerting but the =3D
in-band information should be passed to the calling party rather than =
=3D
local ringing. Finally, 180 (18) and 183 (20) clearly indicate that A =
=3D
should generate local ringing until called party answer or some other =
=3D
change in early media handling. This flow is identical to the original =
=3D
flow except for the addition of the IB indication to four of the =3D
provisional responses in (8), (9), (14) and (15).
------_=3D_NextPart_000_01C31E4F.EA121D72--
_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP

20-May-2003 12:33:45 -0500,13291;000000000000-00000000
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From: "Francois Audet" <audet@nortelnetworks.com>
To: "'Ejzak, Richard P (Richard)'" <ejzak@lucent.com>,
      "'Gonzalo Camarillo Gonzalez (LMF)'" =
<Gonzalo.Camarillo@lmf.ericsson.se>,
      "'sipping@ietf.org'" <sipping@ietf.org>
Cc: "'Jon Peterson (E-mail)'" <jon.peterson@neustar.biz>,
      "'Rohan Mahy (E-mail)'" <rohan@cisco.com>,
      "'Dean Willis (E-mail)'"
                  <dean.willis@softarmor.com>
Subject: RE: [Sipping] Last Call for early media use cases
Date: Tue, 20 May 2003 10:32:21 -0700
MIME-Version: 1.0
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------_=3D_NextPart_001_01C31EF5.C4F625B4
Thanks Richard, this is exactly one of the scenario I had in mind.

Now let's look at a different but very similar scenario (which is the
one that worries me most).

Assume the same exact call flows, but where the calls to C, D and
E are not done in sequence, but in parrallel (i.e., parallel forking).

From A's perspective, the scenario is exactly the same except that it
will receive the provisional responses from B, C, D and E in random
order, and much closer together. Let's assume that they arrive in the
following order (E, C, D), milliseconds apart.

It wouldn't make sense to change the ringing patterns "sequentially"
because they are not sequenced, but parallel. With the current proposal
(and no means to distinguish between parrallel forked calls and=20
sequential calls), we would end-up with D's ringing (out-of-band) and=20
E's answer. If D was busy (in-band), the user would hear "busy" tone=20
followed by an answer, which is not acceptable.

Two solutions exist for that problem:
- That ALL forking entities be B2BUAs and shield this problem from the =
UAs.
    If there is even just one forking proxy out there (making calls in=20
    parallel), we will get the problem. I'm sceptical that we can =
convince
    the whole world NOT to use forking proxies to make parallel calls.
- That forking proxies tag along a "parrallel forking" indication that=20
    will be provided to the called UAs. This indication will tell the UA =
that=20
    in-band ringing (or other tones) in early media shall NOT be used
    because it is useless (early media for answer of course would still =
be
    valid). (It may make sense also to let the UA sending the original =
invite
    know about the forking so it can provide appropriate treatment for =
the
    user: a tone, or display indication).

The second solution is in my opinion the best. And it would be =
compatible
with your proposal for sequential forwarding. Proxies (or B2BUAs) doing
sequential forwarding would NOT put this "parallel forking" indication.
Your proposed in-band call progress indication would solve the problem=20
nicely.

Proxies (or B2BUAs) doing parralel forking would put the "parallel =
forking"
indication. This would prohibit the UAs receiving the INVITE to=20
do in-band call progress tones (i.e., they wouldn't be allowed to =
include
your in-band call progress indication).=20

In combination, the "in-band call progress indication" and the =
"parralel
forking indiction" would solve all the scenarios that were worrying me.

> -----Original Message-----
> From: Ejzak, Richard P (Richard) [mailto:ejzak@lucent.com]=20
> Sent: Monday, May 19, 2003 14:45
> To: 'Gonzalo Camarillo Gonzalez (LMF)'; 'sipping@ietf.org'
> Cc: Jon Peterson (E-mail); Rohan Mahy (E-mail); Dean Willis (E-mail)
> Subject: RE: [Sipping] Last Call for early media use cases
>=20
>=20
> Gonzalo,
> Please excuse my tardiness in responding to your request for=20
> early media use cases. I believe the attachment includes an=20
> example configuration and call flow that clearly illustrates=20
> some problems that need to be solved with the existing SIP=20
> early media procedures.
>=20
> Thanks,
>=20
> Richard Ejzak
> Lucent Technologies
>=20
> -----Original Message-----
> From: Gonzalo Camarillo Gonzalez (LMF)=20
> [mailto:Gonzalo.Camarillo@lmf.ericsson.se]
> Sent: Wednesday,=20
> May 14, 2003 2:13 AM
> To: 'sipping@ietf.org'
> Cc: Jon Peterson (E-mail); Rohan Mahy (E-mail); Dean Willis (E-mail)
> Subject: [Sipping] Last Call for early media use cases
>=20
>=20
> Folks,
>=20
> in the last IETF meeting, we requested early media use cases=20
> that show the need of a "media coming" flag. So far, we have=20
> not received anything.
>=20
> We intend to release a new version of the early media draft=20
> in a week or so. So, that is the time you have to send me=20
> your use cases.
>=20
> Thanks,
>=20
> Gonzalo
> _______________________________________________
> Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current=20
> sip Use sip@ietf.org for new developments of core SIP
>=20
>=20

------_=3D_NextPart_001_01C31EF5.C4F625B4
[Enclosed text/html ]
------_=3D_NextPart_001_01C31EF5.C4F625B4--

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Message-ID: =
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From: "Francois Audet" <audet@nortelnetworks.com>
To: "'Ejzak, Richard P (Richard)'" <ejzak@lucent.com>,
      "'Gonzalo Camarillo Gonzalez (LMF)'" =
<Gonzalo.Camarillo@lmf.ericsson.se>,
      "'sipping@ietf.org'" <sipping@ietf.org>
Cc: "'Jon Peterson (E-mail)'" <jon.peterson@neustar.biz>,
      "'Rohan Mahy (E-mail)'" <rohan@cisco.com>,
      "'Dean Willis (E-mail)'"
                  <dean.willis@softarmor.com>
Subject: RE: [Sipping] Last Call for early media use cases
Date: Tue, 20 May 2003 10:32:21 -0700
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------_=3D_NextPart_001_01C31EF5.C4F625B4
Thanks Richard, this is exactly one of the scenario I had in mind.

Now let's look at a different but very similar scenario (which is the
one that worries me most).

Assume the same exact call flows, but where the calls to C, D and
E are not done in sequence, but in parrallel (i.e., parallel forking).

From A's perspective, the scenario is exactly the same except that it
will receive the provisional responses from B, C, D and E in random
order, and much closer together. Let's assume that they arrive in the
following order (E, C, D), milliseconds apart.

It wouldn't make sense to change the ringing patterns "sequentially"
because they are not sequenced, but parallel. With the current proposal
(and no means to distinguish between parrallel forked calls and=20
sequential calls), we would end-up with D's ringing (out-of-band) and=20
E's answer. If D was busy (in-band), the user would hear "busy" tone=20
followed by an answer, which is not acceptable.

Two solutions exist for that problem:
- That ALL forking entities be B2BUAs and shield this problem from the =
UAs.
    If there is even just one forking proxy out there (making calls in=20
    parallel), we will get the problem. I'm sceptical that we can =
convince
    the whole world NOT to use forking proxies to make parallel calls.
- That forking proxies tag along a "parrallel forking" indication that=20
    will be provided to the called UAs. This indication will tell the UA =
that=20
    in-band ringing (or other tones) in early media shall NOT be used
    because it is useless (early media for answer of course would still =
be
    valid). (It may make sense also to let the UA sending the original =
invite
    know about the forking so it can provide appropriate treatment for =
the
    user: a tone, or display indication).

The second solution is in my opinion the best. And it would be =
compatible
with your proposal for sequential forwarding. Proxies (or B2BUAs) doing
sequential forwarding would NOT put this "parallel forking" indication.
Your proposed in-band call progress indication would solve the problem=20
nicely.

Proxies (or B2BUAs) doing parralel forking would put the "parallel =
forking"
indication. This would prohibit the UAs receiving the INVITE to=20
do in-band call progress tones (i.e., they wouldn't be allowed to =
include
your in-band call progress indication).=20

In combination, the "in-band call progress indication" and the =
"parralel
forking indiction" would solve all the scenarios that were worrying me.

> -----Original Message-----
> From: Ejzak, Richard P (Richard) [mailto:ejzak@lucent.com]=20
> Sent: Monday, May 19, 2003 14:45
> To: 'Gonzalo Camarillo Gonzalez (LMF)'; 'sipping@ietf.org'
> Cc: Jon Peterson (E-mail); Rohan Mahy (E-mail); Dean Willis (E-mail)
> Subject: RE: [Sipping] Last Call for early media use cases
>=20
>=20
> Gonzalo,
> Please excuse my tardiness in responding to your request for=20
> early media use cases. I believe the attachment includes an=20
> example configuration and call flow that clearly illustrates=20
> some problems that need to be solved with the existing SIP=20
> early media procedures.
>=20
> Thanks,
>=20
> Richard Ejzak
> Lucent Technologies
>=20
> -----Original Message-----
> From: Gonzalo Camarillo Gonzalez (LMF)=20
> [mailto:Gonzalo.Camarillo@lmf.ericsson.se]
> Sent: Wednesday,=20
> May 14, 2003 2:13 AM
> To: 'sipping@ietf.org'
> Cc: Jon Peterson (E-mail); Rohan Mahy (E-mail); Dean Willis (E-mail)
> Subject: [Sipping] Last Call for early media use cases
>=20
>=20
> Folks,
>=20
> in the last IETF meeting, we requested early media use cases=20
> that show the need of a "media coming" flag. So far, we have=20
> not received anything.
>=20
> We intend to release a new version of the early media draft=20
> in a week or so. So, that is the time you have to send me=20
> your use cases.
>=20
> Thanks,
>=20
> Gonzalo
> _______________________________________________
> Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current=20
> sip Use sip@ietf.org for new developments of core SIP
>=20
>=20

------_=3D_NextPart_001_01C31EF5.C4F625B4
[Enclosed text/html ]
------_=3D_NextPart_001_01C31EF5.C4F625B4--
_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP


------_=_NextPart_000_01C32350.9CD8ECA6--

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Subject: Re: [Sipping] [Sipping]Questions on draft-ietf-sipping-connect-reuse-reqs-00
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Hi,

On Thursday, May 15, 2003, at 07:23 AM, Nicole Johnson wrote:

> I have a couple of questions about 
> draft-ietf-sipping-connect-reuse-reqs-00.

> 1.  Intended scope of alias parameter.
>
> Is the aim of this draft to enable connections to be reused within a 
> single
> dialog, or across multiple dialogs, i.e. can an alias learned within 
> the
> scope of one dialog be used in another?  Is the alias parameter valid 
> on
> out-of-dialog requests?

across multiple dialogs.  it would work on out-of-dialog requests (ex: 
OPTIONS).
perhaps these should be explicit requirements?

> 2.  Target address fails to authenticate.
>
> If a server receives a request containing the alias parameter and the 
> target
> address of the request fails authentication, how should the server 
> react?

it should not create the alias.

> It could drop the message, respond to the target address with a "not
> authorized" response, or continue processing the request as if the 
> "alias"
> parameter was not present.  Is there a recommended behaviour?

the behavior for handling the request should then do whatever normally 
happens when a request without an alias parameter arrives with failed 
or missing authentication.  frequently this is fine and will result in 
a digest authentication challenge (401 or 407).

hope this makes sense.

thanks,
-rohan

> Thanks,
> Nicole
>
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP

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Subject: Re: [Sipping] WGLC for draft-ietf-sipping-e164-02.txt
From: Cullen Jennings <fluffy@cisco.com>
To: Juha Heinanen <jh@lohi.eng.song.fi>,
        Jon Peterson <jon.peterson@neustar.biz>, Sipping <sipping@ietf.org>
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On 5/15/03 6:06 AM, "Juha Heinanen" <jh@lohi.eng.song.fi> wrote:

> Peterson, Jon writes:
> 
>> I don't
>> think it is too much to ask that SIP applications utilizing ENUM be generous
>> in what they receive with regard to enumservices -  it just doesn't seem
>> costly to be backwards compatible with the original iteration of the ENUM
>> standard.
> 
> check, for example, with cisco and let me know when they accept enum
> replies with both formats.  today they don't which is a very big
> problem. 
> 
> -- juha

If it is the right thing to do, and does not put huge burden on the
implementers (which I don't think this does), I suspect most the vendors
will implement both. My guess would be that the timeline for a vendor to
support both would be very similar to the timeline to support just the new
thing. This is the case for Cisco. I'd rather not have two things if one
would do - it's not so much the implementation issues but ongoing
interoperability and test issues.

I sort of question the idea that this is deployed enough that we have to be
backwards compatible on it. Is this really true? I could be convinced either
way. 

Cullen

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Date: Tue, 20 May 2003 13:15:15 -0700
Subject: Re: [Sipping] Push-To-Talk with SIP
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Hi,

Push to talk can be implemented by merely setting up a session between 
the speaker and the target when the UA becomes available and leaving 
the session open, but only sending audio media over that session when a 
button is pushed.  The session can be verified for liveness by using 
the session-timer, and the target suddenly reappearing after an absence 
can be detected using the registration package.  Perhaps someone could 
write a draft describing this usage?

thanks,
-rohan


On Wednesday, May 14, 2003, at 06:46 PM, ???? ??? wrote:

> Hello,
>  
> Nokia, Siemens and Ericsson have announced that they are developing a 
> push-to-talk application for GSM networks with GPRS and for future 3G 
> networks. They want to use SIP for signaling but I couldn't find any 
> more detailed information.
>  
> Does someone of you know some detail about the PTT signaling with SIP?
> Do they care about arbitration if two people want to talk to the same 
> group at the same time? Are there some drafts or discussions about 
> this topics?
>  
> regards,
> thomas lee.

_______________________________________________
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To: Rohan Mahy <rohan@cisco.com>
Cc: sipping@ietf.org, sipping-admin@ietf.org, thomas.lee@polypix.com
Subject: Re: [Sipping] Push-To-Talk with SIP
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Rohan,
though this method has been mentioned before it does have issues like 
keeping a session active in all billing entities as well
as on UAs. This becomes an issue especially when billing agents may not be 
in media path (in the SIP implementation).
In addition, this would mean that end-phones would necessarily have to be 
able to maintain atleast two calls (one PTT mode,
and the other normal mode ). In addition, needless to mention, in PTT any 
one can burst talk to any other person within a particular
group - which has implications , for example, if A is already 'PTTing' 
with B and C sends a PTT to A, how does A handle the call if 
its already in 'session' ? Another call ? Adhoc mix ?  etc. etc.

For this very reason, some people are also considering 'session less' 
approaches as well, which have their own issues 
wrt. voice packet continuity.

It may be a good idea to discuss an approach and its associated benefits 
and the requirements.

regds
arjun

--
Arjun Roychowdhury @ Hughes Software Systems
11717 Exploration Lane, Germantown MD 20876
(O): 301 212 7860 




Rohan Mahy <rohan@cisco.com>
Sent by: sipping-admin@ietf.org
05/20/2003 04:15 PM

 
        To:     <thomas.lee@polypix.com>
        cc:     <sipping@ietf.org>
        Subject:        Re: [Sipping] Push-To-Talk with SIP



Hi,

Push to talk can be implemented by merely setting up a session between 
the speaker and the target when the UA becomes available and leaving 
the session open, but only sending audio media over that session when a 
button is pushed.  The session can be verified for liveness by using 
the session-timer, and the target suddenly reappearing after an absence 
can be detected using the registration package.  Perhaps someone could 
write a draft describing this usage?

thanks,
-rohan


On Wednesday, May 14, 2003, at 06:46 PM, ???? ??? wrote:

> Hello,
>  
> Nokia, Siemens and Ericsson have announced that they are developing a 
> push-to-talk application for GSM networks with GPRS and for future 3G 
> networks. They want to use SIP for signaling but I couldn't find any 
> more detailed information.
>  
> Does someone of you know some detail about the PTT signaling with SIP?
> Do they care about arbitration if two people want to talk to the same 
> group at the same time? Are there some drafts or discussions about 
> this topics?
>  
> regards,
> thomas lee.

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Gonzalo Camarillo Gonzalez (LMF)"
	 <Gonzalo.Camarillo@lmf.ericsson.se>
To: "'Rohan Mahy'" <rohan@cisco.com>, thomas.lee@polypix.com
Cc: sipping@ietf.org
Subject: RE: [Sipping] Push-To-Talk with SIP
Date: Wed, 21 May 2003 04:46:42 +0200
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Rohan,

a push-to-talk service would include some form of floor control as well.

Gonzalo

-----Original Message-----
From: Rohan Mahy [mailto:rohan@cisco.com]
Sent: Tuesday, May 20, 2003 11:15 PM
To: thomas.lee@polypix.com
Cc: sipping@ietf.org
Subject: Re: [Sipping] Push-To-Talk with SIP



Hi,

Push to talk can be implemented by merely setting up a session between 
the speaker and the target when the UA becomes available and leaving 
the session open, but only sending audio media over that session when a 
button is pushed.  The session can be verified for liveness by using 
the session-timer, and the target suddenly reappearing after an absence 
can be detected using the registration package.  Perhaps someone could 
write a draft describing this usage?

thanks,
-rohan


On Wednesday, May 14, 2003, at 06:46 PM, ???? ??? wrote:

> Hello,
>  
> Nokia, Siemens and Ericsson have announced that they are developing a 
> push-to-talk application for GSM networks with GPRS and for future 3G 
> networks. They want to use SIP for signaling but I couldn't find any 
> more detailed information.
>  
> Does someone of you know some detail about the PTT signaling with SIP?
> Do they care about arbitration if two people want to talk to the same 
> group at the same time? Are there some drafts or discussions about 
> this topics?
>  
> regards,
> thomas lee.

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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To: "Gonzalo Camarillo Gonzalez (LMF)" <Gonzalo.Camarillo@lmf.ericsson.se>
cc: "'Rohan Mahy'" <rohan@cisco.com>, thomas.lee@polypix.com, sipping@ietf.org
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Date: Wed, 21 May 2003 09:07:43 +0530
Subject: RE: [Sipping] Push-To-Talk with SIP
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Is there a definition what this service means? I have looked at Comverse,
Winphoria, Comworks implementation and that is more of Buddylist based
feature where I can see who is available and speak accordingly. If this is
what is Push-To-Talk, I am not sure if we need a new draft.  May be we need
to define this functionality to start with before someone can come with a
draft.
Thanks,
Subir




"Gonzalo Camarillo Gonzalez (LMF)" <Gonzalo.Camarillo@lmf.ericsson.se> on
05/21/2003 08:16:42 AM

To:   "'Rohan Mahy'" <rohan@cisco.com>, thomas.lee@polypix.com
cc:   sipping@ietf.org (bcc: Subir Saha/HSSBLR)

Subject:  RE: [Sipping] Push-To-Talk with SIP




Rohan,

a push-to-talk service would include some form of floor control as well.

Gonzalo

-----Original Message-----
From: Rohan Mahy [mailto:rohan@cisco.com]
Sent: Tuesday, May 20, 2003 11:15 PM
To: thomas.lee@polypix.com
Cc: sipping@ietf.org
Subject: Re: [Sipping] Push-To-Talk with SIP



Hi,

Push to talk can be implemented by merely setting up a session between
the speaker and the target when the UA becomes available and leaving
the session open, but only sending audio media over that session when a
button is pushed.  The session can be verified for liveness by using
the session-timer, and the target suddenly reappearing after an absence
can be detected using the registration package.  Perhaps someone could
write a draft describing this usage?

thanks,
-rohan


On Wednesday, May 14, 2003, at 06:46 PM, ???? ??? wrote:

> Hello,
>
> Nokia, Siemens and Ericsson have announced that they are developing a
> push-to-talk application for GSM networks with GPRS and for future 3G
> networks. They want to use SIP for signaling but I couldn't find any
> more detailed information.
>
> Does someone of you know some detail about the PTT signaling with SIP?
> Do they care about arbitration if two people want to talk to the same
> group at the same time? Are there some drafts or discussions about
> this topics?
>
> regards,
> thomas lee.

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP




_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Gonzalo Camarillo Gonzalez (LMF)"
	 <Gonzalo.Camarillo@lmf.ericsson.se>
To: "'ssubir@hss.hns.com'" <ssubir@hss.hns.com>
Cc: "'Rohan Mahy'" <rohan@cisco.com>, thomas.lee@polypix.com, sipping@ietf.org
Subject: RE: [Sipping] Push-To-Talk with SIP
Date: Wed, 21 May 2003 05:50:46 +0200
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Subir,

as far as I understand, "Push-To-Talk" is a service offered by a specific company. Other companies offering that type of service need to call it something different. All those services have common functionality, but they have differences as well. There is not *a* PTT service definition (right now).

Gonzalo

-----Original Message-----
From: ssubir@hss.hns.com [mailto:ssubir@hss.hns.com]
Sent: Wednesday, May 21, 2003 6:38 AM
To: Gonzalo Camarillo Gonzalez (LMF)
Cc: 'Rohan Mahy'; thomas.lee@polypix.com; sipping@ietf.org
Subject: RE: [Sipping] Push-To-Talk with SIP




Is there a definition what this service means? I have looked at Comverse,
Winphoria, Comworks implementation and that is more of Buddylist based
feature where I can see who is available and speak accordingly. If this is
what is Push-To-Talk, I am not sure if we need a new draft.  May be we need
to define this functionality to start with before someone can come with a
draft.
Thanks,
Subir




"Gonzalo Camarillo Gonzalez (LMF)" <Gonzalo.Camarillo@lmf.ericsson.se> on
05/21/2003 08:16:42 AM

To:   "'Rohan Mahy'" <rohan@cisco.com>, thomas.lee@polypix.com
cc:   sipping@ietf.org (bcc: Subir Saha/HSSBLR)

Subject:  RE: [Sipping] Push-To-Talk with SIP




Rohan,

a push-to-talk service would include some form of floor control as well.

Gonzalo

-----Original Message-----
From: Rohan Mahy [mailto:rohan@cisco.com]
Sent: Tuesday, May 20, 2003 11:15 PM
To: thomas.lee@polypix.com
Cc: sipping@ietf.org
Subject: Re: [Sipping] Push-To-Talk with SIP



Hi,

Push to talk can be implemented by merely setting up a session between
the speaker and the target when the UA becomes available and leaving
the session open, but only sending audio media over that session when a
button is pushed.  The session can be verified for liveness by using
the session-timer, and the target suddenly reappearing after an absence
can be detected using the registration package.  Perhaps someone could
write a draft describing this usage?

thanks,
-rohan


On Wednesday, May 14, 2003, at 06:46 PM, ???? ??? wrote:

> Hello,
>
> Nokia, Siemens and Ericsson have announced that they are developing a
> push-to-talk application for GSM networks with GPRS and for future 3G
> networks. They want to use SIP for signaling but I couldn't find any
> more detailed information.
>
> Does someone of you know some detail about the PTT signaling with SIP?
> Do they care about arbitration if two people want to talk to the same
> group at the same time? Are there some drafts or discussions about
> this topics?
>
> regards,
> thomas lee.

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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To: "Gonzalo Camarillo Gonzalez (LMF)" <Gonzalo.Camarillo@lmf.ericsson.se>
cc: "'Rohan Mahy'" <rohan@cisco.com>, thomas.lee@polypix.com, sipping@ietf.org
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Subject: RE: [Sipping] Push-To-Talk with SIP
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Gonzalo:
In that case, may be, we can start form here - trying to define a Push to
talk service and then see what is missing in SIP/SIMPLE to achive such
functionality!

Subir




"Gonzalo Camarillo Gonzalez (LMF)" <Gonzalo.Camarillo@lmf.ericsson.se> on
05/21/2003 09:20:46 AM

To:   Subir Saha/HSSBLR
cc:   "'Rohan Mahy'" <rohan@cisco.com>, thomas.lee@polypix.com,
      sipping@ietf.org

Subject:  RE: [Sipping] Push-To-Talk with SIP




Subir,

as far as I understand, "Push-To-Talk" is a service offered by a specific
company. Other companies offering that type of service need to call it
something different. All those services have common functionality, but they
have differences as well. There is not *a* PTT service definition (right
now).

Gonzalo

-----Original Message-----
From: ssubir@hss.hns.com [mailto:ssubir@hss.hns.com]
Sent: Wednesday, May 21, 2003 6:38 AM
To: Gonzalo Camarillo Gonzalez (LMF)
Cc: 'Rohan Mahy'; thomas.lee@polypix.com; sipping@ietf.org
Subject: RE: [Sipping] Push-To-Talk with SIP




Is there a definition what this service means? I have looked at Comverse,
Winphoria, Comworks implementation and that is more of Buddylist based
feature where I can see who is available and speak accordingly. If this is
what is Push-To-Talk, I am not sure if we need a new draft.  May be we need
to define this functionality to start with before someone can come with a
draft.
Thanks,
Subir




"Gonzalo Camarillo Gonzalez (LMF)" <Gonzalo.Camarillo@lmf.ericsson.se> on
05/21/2003 08:16:42 AM

To:   "'Rohan Mahy'" <rohan@cisco.com>, thomas.lee@polypix.com
cc:   sipping@ietf.org (bcc: Subir Saha/HSSBLR)

Subject:  RE: [Sipping] Push-To-Talk with SIP




Rohan,

a push-to-talk service would include some form of floor control as well.

Gonzalo

-----Original Message-----
From: Rohan Mahy [mailto:rohan@cisco.com]
Sent: Tuesday, May 20, 2003 11:15 PM
To: thomas.lee@polypix.com
Cc: sipping@ietf.org
Subject: Re: [Sipping] Push-To-Talk with SIP



Hi,

Push to talk can be implemented by merely setting up a session between
the speaker and the target when the UA becomes available and leaving
the session open, but only sending audio media over that session when a
button is pushed.  The session can be verified for liveness by using
the session-timer, and the target suddenly reappearing after an absence
can be detected using the registration package.  Perhaps someone could
write a draft describing this usage?

thanks,
-rohan


On Wednesday, May 14, 2003, at 06:46 PM, ???? ??? wrote:

> Hello,
>
> Nokia, Siemens and Ericsson have announced that they are developing a
> push-to-talk application for GSM networks with GPRS and for future 3G
> networks. They want to use SIP for signaling but I couldn't find any
> more detailed information.
>
> Does someone of you know some detail about the PTT signaling with SIP?
> Do they care about arbitration if two people want to talk to the same
> group at the same time? Are there some drafts or discussions about
> this topics?
>
> regards,
> thomas lee.

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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: [Sipping] [Sip] RE: SIP mobility in WLAN
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Santharam, Arun [NTWK SVCS] writes:
 > 
 > If we follow the SIP implementation on Mobile IP then the SIP stack
 > should not have to bother about the mobility aspect. The SIP stack
 > should be notified of IP changes/handoff information (if relevant for
 > the SIP stack) by the underlying stack. This may prompt a SIP
 > registration and subsequent call flows.

I think it's a little bit more complicated than
this. As with most things, it often makes sense to
have logic at many different layers depending on
the actual situation you find yourself in. For
example, since there isn't route optimization for
IPv4, a reinvite with the new correct CoA would
rid yourself of the dog-leg through the home
agent. But it might complicate QoS reservations
and/or treatment, etc.

Which is a round about way of saying "it depends".

	 Mike
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From: "Santharam, Arun [NTWK SVCS]" <asanth01@sprintspectrum.com>
To: "Rajarshi Chakraborty" <rajarshi_chakraborty@hotmail.com>,
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If we follow the SIP implementation on Mobile IP then the SIP stack
should not have to bother about the mobility aspect. The SIP stack
should be notified of IP changes/handoff information (if relevant for
the SIP stack) by the underlying stack. This may prompt a SIP
registration and subsequent call flows.

My 2 cents ...
arun


> -----Original Message-----
> From: Rajarshi Chakraborty [mailto:rajarshi_chakraborty@hotmail.com] 
> Sent: Sunday, May 18, 2003 11:42 PM
> To: drage@lucent.com; ashishn@mahindrabt.com; Santharam, Arun 
> [NTWK SVCS]; sip@ietf.org
> Cc: sipping@ietf.org
> Subject: SIP mobility in WLAN
> 
> 
> Hi,
> 
> Where can I get call-flows for SIP session management in a 
> wireless LAN 
> (802.11)? 3GPP only addresses the issues for cellular 
> networks. Or are they 
> too trivial to be addressed specially? I am specially interested in 
> situations of handoffs (in this case, the IP changes as soon 
> as the host 
> shifts from the zone around one hotspot to another). Should 
> SIP bother at 
> all to detect IP changes or should the control be completely 
> outside SIP? I 
> mean should it be some other module that has the 
> responsibility of informing 
> the SIP stack in the mobile host about the IP changes, so 
> that the host may 
> then send a re-INVITE with a new SDP but the call still 
> remains active?
> 
> regards
> Rajarshi
> 
> 
> Rajarshi Chakraborty
> B-196, IIT Campus,
> Kharagpur,
> West Bengal, India
> PIN-721302
> 
> 
> 
> 
> 
> ----Original Message Follows----
> From: "Drage, Keith (Keith)" <drage@lucent.com>
> To: "'Ashish Naik'" <ashishn@mahindrabt.com>,   "Santharam, 
> Arun [CC]" 
> <asanth01@sprintspectrum.com>,   "Sip@Ietf. Org" <sip@ietf.org>
> Subject: RE: [Sip] Session Mobility in SIP
> Date: Thu, 8 May 2003 14:42:24 +0100
> 
> Your best starting point to see the architecture is 3GPP TS 
> 23.228, with 
> reference to 3GPP TS 23.002 to see how these entities fit 
> together into the 
> entire 3GPP architecture.
> 
> The SIP protocol application is covered in 3GPP TS 24.229 
> with example flows 
> in 3GPP TS 24.228.
> 
> These documents may be found at:
> 
> ftp://ftp.3gpp.org/Specs/latest/Rel-5/ 
> <ftp://ftp.3gpp.org/Specs/latest/Rel-5/>
> 
> 3GPP uses these specifications over GPRS to provide mobility.
> 
> There is a parallel set of 3GPP2 specifications, 
> substantially of common 
> text, that uses the same architecture over mobile IP, instead 
> of GPRS. These 
> documents are currently in 3GPP2 comment and review.
> 
> regards
> 
> Keith
> 
> 
> 
> 
> Keith Drage
> Lucent Technologies
> Tel: +44 1793 776249
> Email: drage@lucent.com
> 
> -----Original Message-----
> From: Ashish Naik [mailto:ashishn@mahindrabt.com]
> Sent: 08 May 2003 06:23
> To: Santharam, Arun [CC]; Sip@Ietf. Org
> Subject: RE: [Sip] Session Mobility in SIP
> 
> 
> ok. Is there any specific document on IETF and 3GPP that can 
> explain how it 
> is being addressed  ?
> 
> 
> Ragards,
> Ashish Naik
> CoE Mobile Computing
> -----------------------------------
> Phone 4018100 Ext 3070
> 
> 
> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf 
> Of Santharam, 
> Arun [CC]
> Sent: Wednesday, April 09, 2003 9:42 PM
> To: Ashish Naik; Sip@Ietf. Org
> Subject: RE: [Sip] Session Mobility in SIP
> 
> 
> 
> Ashish,
> 
> 
> 
> Please find comments inline.
> 
> 
> 
> Thank you
> 
> arun
> 
> 
> 
> -----Original Message-----
> From: Ashish Naik [mailto:ashishn@mahindrabt.com]
> Sent: Wednesday, April 09, 2003 4:01 AM
> To: Sip@Ietf. Org
> Subject: [Sip] Session Mobility in SIP
> 
> 
> 
> How is SIP going to address session mobility ? Is there any 
> initiative taken 
> up for this ?
> 
> 
> 
>  > This has been addressed in 3GPP standards. Please refer to 
> the 3GPP All 
> IP architecture specifications.
> 
> 
> 
> SIP is said to be a better alternative than Mobile IP. Can I 
> have some 
> latest development document references ?
> 
> 
> 
> 
> 
>  >  SIP is not going to replace Mobile IP. It is going to 
> leverage Mobile IP 
> for session management in 3G wireless networks.
> 
> 
> 
> 
> 
> Regards,
> Ashish Naik
> -------------------
> CoE Mobile Computing
> Phone: 4018100 Ext 3070
> 
> 
> 
> 
> *********************************************************
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Subject: RE: [Sipping] Push-To-Talk with SIP
Date: Wed, 21 May 2003 15:52:36 +0300
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Hi Rohan,

I think this is one way to look at the push to talk feature, but maybe not exactly "mainstream". Most of the push to talk services I'm aware of are similar to a walkie talkie service, where a group of people are participating, out of which only one at a time can send media to the others. "Requesting the floor" in this case is usually done with the "tangent" button.

So in my opinion, push to talk is a conference, with a specific type of conference and media policy associated with it. As such, I don't think it's the IETF's business to define those policies but provide tools for setting/manipulating them, which I guess the conference DT is already doing. 

	-Aki

 > -----Original Message-----
 > From: ext Rohan Mahy [mailto:rohan@cisco.com]
 > Sent: 20 May, 2003 23:15
 > To: thomas.lee@polypix.com
 > Cc: sipping@ietf.org
 > Subject: Re: [Sipping] Push-To-Talk with SIP
 > 
 > 
 > 
 > Hi,
 > 
 > Push to talk can be implemented by merely setting up a 
 > session between 
 > the speaker and the target when the UA becomes available and leaving 
 > the session open, but only sending audio media over that 
 > session when a 
 > button is pushed.  The session can be verified for liveness by using 
 > the session-timer, and the target suddenly reappearing after 
 > an absence 
 > can be detected using the registration package.  Perhaps 
 > someone could 
 > write a draft describing this usage?
 > 
 > thanks,
 > -rohan
 > 
 > 
 > On Wednesday, May 14, 2003, at 06:46 PM, ???? ??? wrote:
 > 
 > > Hello,
 > >  
 > > Nokia, Siemens and Ericsson have announced that they are 
 > developing a 
 > > push-to-talk application for GSM networks with GPRS and 
 > for future 3G 
 > > networks. They want to use SIP for signaling but I 
 > couldn't find any 
 > > more detailed information.
 > >  
 > > Does someone of you know some detail about the PTT 
 > signaling with SIP?
 > > Do they care about arbitration if two people want to talk 
 > to the same 
 > > group at the same time? Are there some drafts or discussions about 
 > > this topics?
 > >  
 > > regards,
 > > thomas lee.
 > 
 > _______________________________________________
 > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
 > This list is for NEW development of the application of SIP
 > Use sip-implementors@cs.columbia.edu for questions on current sip
 > Use sip@ietf.org for new developments of core SIP
 > 
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Hi,

There has been discussion about a possible (SIP?) Conferencing BOF in the forthcoming Vienna IETF meeting, so that Conferencing related work from SIPPING and MMUSIC would be trasfered to a new WG. Is this already definite or just a proposal? Is anyone working on the charter proposal for this?

Markus 
_______________________________________________
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This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



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To: aki.niemi@nokia.com
Cc: rohan@cisco.com, sipping@ietf.org, sipping-admin@ietf.org,
        thomas.lee@polypix.com
Subject: RE: [Sipping] Push-To-Talk with SIP
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From: "Arjun Roychowdhury" <aroychow@hns.com>
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I think to avoid confusion, it may be better to reword this is a 'session 
less two way radio service',
with session less being the key item. I believe this was the original 
question of the poster.

Whether this particular approach makes sense to be done in SIP and 
especially in the purview of the IETF  is another question.

regds
arjun

--
Arjun Roychowdhury @ Hughes Software Systems
11717 Exploration Lane, Germantown MD 20876
(O): 301 212 7860 




aki.niemi@nokia.com
Sent by: sipping-admin@ietf.org
05/21/2003 08:52 AM

 
        To:     <rohan@cisco.com>, <thomas.lee@polypix.com>
        cc:     <sipping@ietf.org>
        Subject:        RE: [Sipping] Push-To-Talk with SIP


Hi Rohan,

I think this is one way to look at the push to talk feature, but maybe not 
exactly "mainstream". Most of the push to talk services I'm aware of are 
similar to a walkie talkie service, where a group of people are 
participating, out of which only one at a time can send media to the 
others. "Requesting the floor" in this case is usually done with the 
"tangent" button.

So in my opinion, push to talk is a conference, with a specific type of 
conference and media policy associated with it. As such, I don't think 
it's the IETF's business to define those policies but provide tools for 
setting/manipulating them, which I guess the conference DT is already 
doing. 

                 -Aki

 > -----Original Message-----
 > From: ext Rohan Mahy [mailto:rohan@cisco.com]
 > Sent: 20 May, 2003 23:15
 > To: thomas.lee@polypix.com
 > Cc: sipping@ietf.org
 > Subject: Re: [Sipping] Push-To-Talk with SIP
 > 
 > 
 > 
 > Hi,
 > 
 > Push to talk can be implemented by merely setting up a 
 > session between 
 > the speaker and the target when the UA becomes available and leaving 
 > the session open, but only sending audio media over that 
 > session when a 
 > button is pushed.  The session can be verified for liveness by using 
 > the session-timer, and the target suddenly reappearing after 
 > an absence 
 > can be detected using the registration package.  Perhaps 
 > someone could 
 > write a draft describing this usage?
 > 
 > thanks,
 > -rohan
 > 
 > 
 > On Wednesday, May 14, 2003, at 06:46 PM, ???? ??? wrote:
 > 
 > > Hello,
 > >  
 > > Nokia, Siemens and Ericsson have announced that they are 
 > developing a 
 > > push-to-talk application for GSM networks with GPRS and 
 > for future 3G 
 > > networks. They want to use SIP for signaling but I 
 > couldn't find any 
 > > more detailed information.
 > >  
 > > Does someone of you know some detail about the PTT 
 > signaling with SIP?
 > > Do they care about arbitration if two people want to talk 
 > to the same 
 > > group at the same time? Are there some drafts or discussions about 
 > > this topics?
 > >  
 > > regards,
 > > thomas lee.
 > 
 > _______________________________________________
 > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
 > This list is for NEW development of the application of SIP
 > Use sip-implementors@cs.columbia.edu for questions on current sip
 > Use sip@ietf.org for new developments of core SIP
 > 
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Wed, 21 May 2003 11:33:25 -0500
From: Alan Johnston <alan.johnston@mci.com>
Subject: Re: [Sipping] Conferencing BOF/WG
In-reply-to: <E392EEA75EC5F54AB75229B693B1B6A70236807F@esebe018.ntc.noki a.com>
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Hi Markus,

Yes, a BOF is being planned but is not on the agenda yet.  The documents 
which have not become SIPPING WG documents from the SIPPING Design Team 
would be the basis (http://ee.wustl.edu/~alan/sip-conf)

When we have more news including a mailing list for discussion of the 
proposed charter and documents, we'll announce on the SIP, SIPPING, and 
MMUSIC lists.

Thanks,
Alan Johnston
MCI

At 05:01 PM 5/21/2003 +0300, Markus.Isomaki@nokia.com wrote:
>Hi,
>
>There has been discussion about a possible (SIP?) Conferencing BOF in the 
>forthcoming Vienna IETF meeting, so that Conferencing related work from 
>SIPPING and MMUSIC would be trasfered to a new WG. Is this already 
>definite or just a proposal? Is anyone working on the charter proposal for 
>this?
>
>Markus
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: RE: [Sipping] Push-To-Talk with SIP
Date: Thu, 22 May 2003 12:32:26 +0300
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Hi Arjun,

I don't really understand what you mean by 'session less', could you elaborate a bit? Do you mean to say there is no application state maintained at the endpoints? 

And to me, the original question seemed to be whether there have been any drafts or discussions w.r.t. push to talk and SIP in the IETF. And the enswer to that would have to be no, but the conferencing work in SIPPING seems to be covering related issues, e.g., floor control, and media policy.

Regards,	
Aki


 > -----Original Message-----
 > From: ext Arjun Roychowdhury [mailto:aroychow@hns.com]
 > Sent: 21 May, 2003 17:15
 > To: Niemi Aki (NMP/Helsinki)
 > Cc: rohan@cisco.com; sipping@ietf.org; sipping-admin@ietf.org;
 > thomas.lee@polypix.com
 > Subject: RE: [Sipping] Push-To-Talk with SIP
 > 
 > 
 > I think to avoid confusion, it may be better to reword this 
 > is a 'session 
 > less two way radio service',
 > with session less being the key item. I believe this was the 
 > original 
 > question of the poster.
 > 
 > Whether this particular approach makes sense to be done in SIP and 
 > especially in the purview of the IETF  is another question.
 > 
 > regds
 > arjun
 > 
 > --
 > Arjun Roychowdhury @ Hughes Software Systems
 > 11717 Exploration Lane, Germantown MD 20876
 > (O): 301 212 7860 
 > 
 > 
 > 
 > 
 > aki.niemi@nokia.com
 > Sent by: sipping-admin@ietf.org
 > 05/21/2003 08:52 AM
 > 
 >  
 >         To:     <rohan@cisco.com>, <thomas.lee@polypix.com>
 >         cc:     <sipping@ietf.org>
 >         Subject:        RE: [Sipping] Push-To-Talk with SIP
 > 
 > 
 > Hi Rohan,
 > 
 > I think this is one way to look at the push to talk feature, 
 > but maybe not 
 > exactly "mainstream". Most of the push to talk services I'm 
 > aware of are 
 > similar to a walkie talkie service, where a group of people are 
 > participating, out of which only one at a time can send media to the 
 > others. "Requesting the floor" in this case is usually done with the 
 > "tangent" button.
 > 
 > So in my opinion, push to talk is a conference, with a 
 > specific type of 
 > conference and media policy associated with it. As such, I 
 > don't think 
 > it's the IETF's business to define those policies but 
 > provide tools for 
 > setting/manipulating them, which I guess the conference DT 
 > is already 
 > doing. 
 > 
 >                  -Aki
 > 
 >  > -----Original Message-----
 >  > From: ext Rohan Mahy [mailto:rohan@cisco.com]
 >  > Sent: 20 May, 2003 23:15
 >  > To: thomas.lee@polypix.com
 >  > Cc: sipping@ietf.org
 >  > Subject: Re: [Sipping] Push-To-Talk with SIP
 >  > 
 >  > 
 >  > 
 >  > Hi,
 >  > 
 >  > Push to talk can be implemented by merely setting up a 
 >  > session between 
 >  > the speaker and the target when the UA becomes available 
 > and leaving 
 >  > the session open, but only sending audio media over that 
 >  > session when a 
 >  > button is pushed.  The session can be verified for 
 > liveness by using 
 >  > the session-timer, and the target suddenly reappearing after 
 >  > an absence 
 >  > can be detected using the registration package.  Perhaps 
 >  > someone could 
 >  > write a draft describing this usage?
 >  > 
 >  > thanks,
 >  > -rohan
 >  > 
 >  > 
 >  > On Wednesday, May 14, 2003, at 06:46 PM, ???? ??? wrote:
 >  > 
 >  > > Hello,
 >  > >  
 >  > > Nokia, Siemens and Ericsson have announced that they are 
 >  > developing a 
 >  > > push-to-talk application for GSM networks with GPRS and 
 >  > for future 3G 
 >  > > networks. They want to use SIP for signaling but I 
 >  > couldn't find any 
 >  > > more detailed information.
 >  > >  
 >  > > Does someone of you know some detail about the PTT 
 >  > signaling with SIP?
 >  > > Do they care about arbitration if two people want to talk 
 >  > to the same 
 >  > > group at the same time? Are there some drafts or 
 > discussions about 
 >  > > this topics?
 >  > >  
 >  > > regards,
 >  > > thomas lee.
 >  > 
 >  > _______________________________________________
 >  > Sipping mailing list  
 > https://www1.ietf.org/mailman/listinfo/sipping
 >  > This list is for NEW development of the application of SIP
 >  > Use sip-implementors@cs.columbia.edu for questions on current sip
 >  > Use sip@ietf.org for new developments of core SIP
 >  > 
 > _______________________________________________
 > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
 > This list is for NEW development of the application of SIP
 > Use sip-implementors@cs.columbia.edu for questions on current sip
 > Use sip@ietf.org for new developments of core SIP
 > 
 > 
 > 
 > 
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: Nicole Johnson <NJ2@dataconnection.com>
To: "'Rohan Mahy'" <rohan@cisco.com>
Cc: "'sipping@ietf.org'" <sipping@ietf.org>
Subject: RE: [Sipping] [Sipping]Questions on draft-ietf-sipping-connect-re
	use-reqs-00
Date: Thu, 22 May 2003 13:12:48 +0100
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Thanks for the clarification.

I have two comments about the second security mechanism discussed in section
4.1 of this draft.

1. The draft states that, for a server to accept an alias, the target
address of that alias must previously have been registered with the server.
However, this mechanism is only as secure as the REGISTER itself.  For
example, if the REGISTER is not integrity-checked, a man-in-the-middle could
alter the contact address in the REGISTER.

I think the draft should recommend that the REGISTER also be
integrity-checked using S/MIME and a sipfrag bodypart containing the
Contact.  I would be happy to propose some text.  Alternatively, if I have
misunderstood the proposed mechanism, please let me know where I have gone
wrong.

2.  It is not clear how this mechanism could be used to reuse connections
between two proxies in a secure manner.  Would each proxy have to REGISTER
its address with the other?

Thanks,
Nicole

-----Original Message-----
From: Rohan Mahy [mailto:rohan@cisco.com]
Sent: Tuesday, May 20, 2003 9:09 PM
To: Nicole Johnson
Cc: 'sipping@ietf.org'
Subject: Re: [Sipping] [Sipping]Questions on
draft-ietf-sipping-connect-reuse-reqs-00


Hi,

On Thursday, May 15, 2003, at 07:23 AM, Nicole Johnson wrote:

> I have a couple of questions about 
> draft-ietf-sipping-connect-reuse-reqs-00.

> 1.  Intended scope of alias parameter.
>
> Is the aim of this draft to enable connections to be reused within a 
> single
> dialog, or across multiple dialogs, i.e. can an alias learned within 
> the
> scope of one dialog be used in another?  Is the alias parameter valid 
> on
> out-of-dialog requests?

across multiple dialogs.  it would work on out-of-dialog requests (ex: 
OPTIONS).
perhaps these should be explicit requirements?

> 2.  Target address fails to authenticate.
>
> If a server receives a request containing the alias parameter and the 
> target
> address of the request fails authentication, how should the server 
> react?

it should not create the alias.

> It could drop the message, respond to the target address with a "not
> authorized" response, or continue processing the request as if the 
> "alias"
> parameter was not present.  Is there a recommended behaviour?

the behavior for handling the request should then do whatever normally 
happens when a request without an alias parameter arrives with failed 
or missing authentication.  frequently this is fine and will result in 
a digest authentication challenge (401 or 407).

hope this makes sense.

thanks,
-rohan

> Thanks,
> Nicole
>
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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To: aki.niemi@nokia.com
Cc: aroychow@hns.com, rohan@cisco.com, sipping@ietf.org,
        sipping-admin@ietf.org, thomas.lee@polypix.com
Subject: RE: [Sipping] Push-To-Talk with SIP
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Aki,
By 'session less' I mean not having an INVITE-OK-ACK to actually establish =

the call before you could talk. Application state would be maintained
during one 'push-release' operation and forgotten after that (except for=20
the administrative states like which was the gruop set I was talking to).

Different algorithms have been suggested for this 'voice over IM' =3D> some=
=20
are also patented @ the uspto (I think someone had posted the patent no.
in the impp archive some time ago)

Anyway, therefore, I realize this may not be something that the ietf sip=20
grp wants to take up unless there is a compelling reason=20
go into into this 2 way radio service. With respect to the original Q, I=20
agree that one could conceptualize that a call in session with RTP
bursts and conferencing thrown in could look like a '2 way radio service'=20
=3D but due to the nature of its persistance of state for a much longer
time than currently known implementations not using SIP, the charging=20
model and complexities would likely not be the same, for what ever its=20
worth.

regds
arjun

--
Arjun Roychowdhury @ Hughes Software Systems
11717 Exploration Lane, Germantown MD 20876
(O): 301 212 7860=20




aki.niemi@nokia.com
Sent by: sipping-admin@ietf.org
05/22/2003 05:32 AM

=20
        To:     <aroychow@hns.com>
        cc:     <rohan@cisco.com>, <sipping@ietf.org>, <sipping-admin@ietf.=
org>,=20
<thomas.lee@polypix.com>
        Subject:        RE: [Sipping] Push-To-Talk with SIP


Hi Arjun,

I don't really understand what you mean by 'session less', could you=20
elaborate a bit? Do you mean to say there is no application state=20
maintained at the endpoints?=20

And to me, the original question seemed to be whether there have been any=20
drafts or discussions w.r.t. push to talk and SIP in the IETF. And the=20
enswer to that would have to be no, but the conferencing work in SIPPING=20
seems to be covering related issues, e.g., floor control, and media=20
policy.

Regards,=20
Aki


 > -----Original Message-----
 > From: ext Arjun Roychowdhury [mailto:aroychow@hns.com]
 > Sent: 21 May, 2003 17:15
 > To: Niemi Aki (NMP/Helsinki)
 > Cc: rohan@cisco.com; sipping@ietf.org; sipping-admin@ietf.org;
 > thomas.lee@polypix.com
 > Subject: RE: [Sipping] Push-To-Talk with SIP
 >=20
 >=20
 > I think to avoid confusion, it may be better to reword this=20
 > is a 'session=20
 > less two way radio service',
 > with session less being the key item. I believe this was the=20
 > original=20
 > question of the poster.
 >=20
 > Whether this particular approach makes sense to be done in SIP and=20
 > especially in the purview of the IETF  is another question.
 >=20
 > regds
 > arjun
 >=20
 > --
 > Arjun Roychowdhury @ Hughes Software Systems
 > 11717 Exploration Lane, Germantown MD 20876
 > (O): 301 212 7860=20
 >=20
 >=20
 >=20
 >=20
 > aki.niemi@nokia.com
 > Sent by: sipping-admin@ietf.org
 > 05/21/2003 08:52 AM
 >=20
 >=20
 >         To:     <rohan@cisco.com>, <thomas.lee@polypix.com>
 >         cc:     <sipping@ietf.org>
 >         Subject:        RE: [Sipping] Push-To-Talk with SIP
 >=20
 >=20
 > Hi Rohan,
 >=20
 > I think this is one way to look at the push to talk feature,=20
 > but maybe not=20
 > exactly "mainstream". Most of the push to talk services I'm=20
 > aware of are=20
 > similar to a walkie talkie service, where a group of people are=20
 > participating, out of which only one at a time can send media to the=20
 > others. "Requesting the floor" in this case is usually done with the=20
 > "tangent" button.
 >=20
 > So in my opinion, push to talk is a conference, with a=20
 > specific type of=20
 > conference and media policy associated with it. As such, I=20
 > don't think=20
 > it's the IETF's business to define those policies but=20
 > provide tools for=20
 > setting/manipulating them, which I guess the conference DT=20
 > is already=20
 > doing.=20
 >=20
 >                  -Aki
 >=20
 >  > -----Original Message-----
 >  > From: ext Rohan Mahy [mailto:rohan@cisco.com]
 >  > Sent: 20 May, 2003 23:15
 >  > To: thomas.lee@polypix.com
 >  > Cc: sipping@ietf.org
 >  > Subject: Re: [Sipping] Push-To-Talk with SIP
 >  >=20
 >  >=20
 >  >=20
 >  > Hi,
 >  >=20
 >  > Push to talk can be implemented by merely setting up a=20
 >  > session between=20
 >  > the speaker and the target when the UA becomes available=20
 > and leaving=20
 >  > the session open, but only sending audio media over that=20
 >  > session when a=20
 >  > button is pushed.  The session can be verified for=20
 > liveness by using=20
 >  > the session-timer, and the target suddenly reappearing after=20
 >  > an absence=20
 >  > can be detected using the registration package.  Perhaps=20
 >  > someone could=20
 >  > write a draft describing this usage?
 >  >=20
 >  > thanks,
 >  > -rohan
 >  >=20
 >  >=20
 >  > On Wednesday, May 14, 2003, at 06:46 PM, ???? ??? wrote:
 >  >=20
 >  > > Hello,
 >  > > =A0
 >  > > Nokia, Siemens and Ericsson have announced that they are=20
 >  > developing a=20
 >  > > push-to-talk application for GSM networks with GPRS and=20
 >  > for future 3G=20
 >  > > networks. They want to use SIP for signaling but I=20
 >  > couldn't find any=20
 >  > > more detailed information.
 >  > > =A0
 >  > > Does someone of you know some detail about the PTT=20
 >  > signaling with SIP?
 >  > > Do they care about arbitration if two people want to talk=20
 >  > to the same=20
 >  > > group at the same time? Are there some drafts or=20
 > discussions about=20
 >  > > this topics?
 >  > > =A0
 >  > > regards,
 >  > > thomas lee.
 >  >=20
 >  > =5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F
 >  > Sipping mailing list=20
 > https://www1.ietf.org/mailman/listinfo/sipping
 >  > This list is for NEW development of the application of SIP
 >  > Use sip-implementors@cs.columbia.edu for questions on current sip
 >  > Use sip@ietf.org for new developments of core SIP
 >  >=20
 > =5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F
 > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
 > This list is for NEW development of the application of SIP
 > Use sip-implementors@cs.columbia.edu for questions on current sip
 > Use sip@ietf.org for new developments of core SIP
 >=20
 >=20
 >=20
 >=20
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



--=_alternative 0043997085256D2E_=
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable


<br><font size=3D2 face=3D"sans-serif">Aki,</font>
<br><font size=3D2 face=3D"sans-serif">By 'session less' I mean not having =
an INVITE-OK-ACK to actually establish the call before you could talk. Appl=
ication state would be maintained</font>
<br><font size=3D2 face=3D"sans-serif">during one 'push-release' operation =
and forgotten after that (except for the administrative states like which w=
as the gruop set I was talking to).</font>
<br>
<br><font size=3D2 face=3D"sans-serif">Different algorithms have been sugge=
sted for this 'voice over IM' =3D&gt; some are also patented @ the uspto (I=
 think someone had posted the patent no.</font>
<br><font size=3D2 face=3D"sans-serif">in the impp archive some time ago)</=
font>
<br>
<br><font size=3D2 face=3D"sans-serif">Anyway, therefore, I realize this ma=
y not be something that the ietf sip grp wants to take up unless there is a=
 compelling reason </font>
<br><font size=3D2 face=3D"sans-serif">go into into this 2 way radio servic=
e. With respect to the original Q, I agree that one could conceptualize tha=
t a call in session with RTP</font>
<br><font size=3D2 face=3D"sans-serif">bursts and conferencing thrown in co=
uld look like a '2 way radio service' =3D but due to the nature of its pers=
istance of state for a much longer</font>
<br><font size=3D2 face=3D"sans-serif">time than currently known implementa=
tions not using SIP, the charging model and complexities would likely not b=
e the same, for what ever its worth.</font>
<br>
<br><font size=3D2 face=3D"sans-serif">regds</font>
<br><font size=3D2 face=3D"sans-serif">arjun</font>
<br>
<br><font size=3D2 face=3D"sans-serif">--<br>
Arjun Roychowdhury @ Hughes Software Systems<br>
11717 Exploration Lane, Germantown MD 20876<br>
(O): 301 212 7860 &nbsp;</font>
<br>
<br>
<br>
<table width=3D100%>
<tr valign=3Dtop>
<td>
<td><font size=3D1 face=3D"sans-serif"><b>aki.niemi@nokia.com</b></font>
<br><font size=3D1 face=3D"sans-serif">Sent by: sipping-admin@ietf.org</fon=
t>
<p><font size=3D1 face=3D"sans-serif">05/22/2003 05:32 AM</font>
<br>
<td><font size=3D1 face=3D"Arial">&nbsp; &nbsp; &nbsp; &nbsp; </font>
<br><font size=3D1 face=3D"sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; To: &nbs=
p; &nbsp; &nbsp; &nbsp;&lt;aroychow@hns.com&gt;</font>
<br><font size=3D1 face=3D"sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; cc: &nbs=
p; &nbsp; &nbsp; &nbsp;&lt;rohan@cisco.com&gt;, &lt;sipping@ietf.org&gt;, &=
lt;sipping-admin@ietf.org&gt;, &lt;thomas.lee@polypix.com&gt;</font>
<br><font size=3D1 face=3D"sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Subject:=
 &nbsp; &nbsp; &nbsp; &nbsp;RE: [Sipping] Push-To-Talk with SIP</font></tab=
le>
<br>
<br>
<br><font size=3D2 face=3D"Courier New">Hi Arjun,<br>
<br>
I don't really understand what you mean by 'session less', could you elabor=
ate a bit? Do you mean to say there is no application state maintained at t=
he endpoints? <br>
<br>
And to me, the original question seemed to be whether there have been any d=
rafts or discussions w.r.t. push to talk and SIP in the IETF. And the enswe=
r to that would have to be no, but the conferencing work in SIPPING seems t=
o be covering related issues, e.g., floor control, and media policy.<br>
<br>
Regards, &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; <br>
Aki<br>
<br>
<br>
 &gt; -----Original Message-----<br>
 &gt; From: ext Arjun Roychowdhury [mailto:aroychow@hns.com]<br>
 &gt; Sent: 21 May, 2003 17:15<br>
 &gt; To: Niemi Aki (NMP/Helsinki)<br>
 &gt; Cc: rohan@cisco.com; sipping@ietf.org; sipping-admin@ietf.org;<br>
 &gt; thomas.lee@polypix.com<br>
 &gt; Subject: RE: [Sipping] Push-To-Talk with SIP<br>
 &gt; <br>
 &gt; <br>
 &gt; I think to avoid confusion, it may be better to reword this <br>
 &gt; is a 'session <br>
 &gt; less two way radio service',<br>
 &gt; with session less being the key item. I believe this was the <br>
 &gt; original <br>
 &gt; question of the poster.<br>
 &gt; <br>
 &gt; Whether this particular approach makes sense to be done in SIP and <b=
r>
 &gt; especially in the purview of the IETF &nbsp;is another question.<br>
 &gt; <br>
 &gt; regds<br>
 &gt; arjun<br>
 &gt; <br>
 &gt; --<br>
 &gt; Arjun Roychowdhury @ Hughes Software Systems<br>
 &gt; 11717 Exploration Lane, Germantown MD 20876<br>
 &gt; (O): 301 212 7860 <br>
 &gt; <br>
 &gt; <br>
 &gt; <br>
 &gt; <br>
 &gt; aki.niemi@nokia.com<br>
 &gt; Sent by: sipping-admin@ietf.org<br>
 &gt; 05/21/2003 08:52 AM<br>
 &gt; <br>
 &gt; &nbsp;<br>
 &gt; &nbsp; &nbsp; &nbsp; &nbsp; To: &nbsp; &nbsp; &lt;rohan@cisco.com&gt;=
, &lt;thomas.lee@polypix.com&gt;<br>
 &gt; &nbsp; &nbsp; &nbsp; &nbsp; cc: &nbsp; &nbsp; &lt;sipping@ietf.org&gt=
;<br>
 &gt; &nbsp; &nbsp; &nbsp; &nbsp; Subject: &nbsp; &nbsp; &nbsp; &nbsp;RE: [=
Sipping] Push-To-Talk with SIP<br>
 &gt; <br>
 &gt; <br>
 &gt; Hi Rohan,<br>
 &gt; <br>
 &gt; I think this is one way to look at the push to talk feature, <br>
 &gt; but maybe not <br>
 &gt; exactly &quot;mainstream&quot;. Most of the push to talk services I'm=
 <br>
 &gt; aware of are <br>
 &gt; similar to a walkie talkie service, where a group of people are <br>
 &gt; participating, out of which only one at a time can send media to the =
<br>
 &gt; others. &quot;Requesting the floor&quot; in this case is usually done=
 with the <br>
 &gt; &quot;tangent&quot; button.<br>
 &gt; <br>
 &gt; So in my opinion, push to talk is a conference, with a <br>
 &gt; specific type of <br>
 &gt; conference and media policy associated with it. As such, I <br>
 &gt; don't think <br>
 &gt; it's the IETF's business to define those policies but <br>
 &gt; provide tools for <br>
 &gt; setting/manipulating them, which I guess the conference DT <br>
 &gt; is already <br>
 &gt; doing. <br>
 &gt; <br>
 &gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;-Aki<br>
 &gt; <br>
 &gt; &nbsp;&gt; -----Original Message-----<br>
 &gt; &nbsp;&gt; From: ext Rohan Mahy [mailto:rohan@cisco.com]<br>
 &gt; &nbsp;&gt; Sent: 20 May, 2003 23:15<br>
 &gt; &nbsp;&gt; To: thomas.lee@polypix.com<br>
 &gt; &nbsp;&gt; Cc: sipping@ietf.org<br>
 &gt; &nbsp;&gt; Subject: Re: [Sipping] Push-To-Talk with SIP<br>
 &gt; &nbsp;&gt; <br>
 &gt; &nbsp;&gt; <br>
 &gt; &nbsp;&gt; <br>
 &gt; &nbsp;&gt; Hi,<br>
 &gt; &nbsp;&gt; <br>
 &gt; &nbsp;&gt; Push to talk can be implemented by merely setting up a <br>
 &gt; &nbsp;&gt; session between <br>
 &gt; &nbsp;&gt; the speaker and the target when the UA becomes available <=
br>
 &gt; and leaving <br>
 &gt; &nbsp;&gt; the session open, but only sending audio media over that <=
br>
 &gt; &nbsp;&gt; session when a <br>
 &gt; &nbsp;&gt; button is pushed. &nbsp;The session can be verified for <b=
r>
 &gt; liveness by using <br>
 &gt; &nbsp;&gt; the session-timer, and the target suddenly reappearing aft=
er <br>
 &gt; &nbsp;&gt; an absence <br>
 &gt; &nbsp;&gt; can be detected using the registration package. &nbsp;Perh=
aps <br>
 &gt; &nbsp;&gt; someone could <br>
 &gt; &nbsp;&gt; write a draft describing this usage?<br>
 &gt; &nbsp;&gt; <br>
 &gt; &nbsp;&gt; thanks,<br>
 &gt; &nbsp;&gt; -rohan<br>
 &gt; &nbsp;&gt; <br>
 &gt; &nbsp;&gt; <br>
 &gt; &nbsp;&gt; On Wednesday, May 14, 2003, at 06:46 PM, ???? ??? wrote:<b=
r>
 &gt; &nbsp;&gt; <br>
 &gt; &nbsp;&gt; &gt; Hello,<br>
 &gt; &nbsp;&gt; &gt; =A0<br>
 &gt; &nbsp;&gt; &gt; Nokia, Siemens and Ericsson have announced that they =
are <br>
 &gt; &nbsp;&gt; developing a <br>
 &gt; &nbsp;&gt; &gt; push-to-talk application for GSM networks with GPRS a=
nd <br>
 &gt; &nbsp;&gt; for future 3G <br>
 &gt; &nbsp;&gt; &gt; networks. They want to use SIP for signaling but I <b=
r>
 &gt; &nbsp;&gt; couldn't find any <br>
 &gt; &nbsp;&gt; &gt; more detailed information.<br>
 &gt; &nbsp;&gt; &gt; =A0<br>
 &gt; &nbsp;&gt; &gt; Does someone of you know some detail about the PTT <b=
r>
 &gt; &nbsp;&gt; signaling with SIP?<br>
 &gt; &nbsp;&gt; &gt; Do they care about arbitration if two people want to =
talk <br>
 &gt; &nbsp;&gt; to the same <br>
 &gt; &nbsp;&gt; &gt; group at the same time? Are there some drafts or <br>
 &gt; discussions about <br>
 &gt; &nbsp;&gt; &gt; this topics?<br>
 &gt; &nbsp;&gt; &gt; =A0<br>
 &gt; &nbsp;&gt; &gt; regards,<br>
 &gt; &nbsp;&gt; &gt; thomas lee.<br>
 &gt; &nbsp;&gt; <br>
 &gt; &nbsp;&gt; =5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F<br>
 &gt; &nbsp;&gt; Sipping mailing list &nbsp;<br>
 &gt; https://www1.ietf.org/mailman/listinfo/sipping<br>
 &gt; &nbsp;&gt; This list is for NEW development of the application of SIP=
<br>
 &gt; &nbsp;&gt; Use sip-implementors@cs.columbia.edu for questions on curr=
ent sip<br>
 &gt; &nbsp;&gt; Use sip@ietf.org for new developments of core SIP<br>
 &gt; &nbsp;&gt; <br>
 &gt; =5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F<br>
 &gt; Sipping mailing list &nbsp;https://www1.ietf.org/mailman/listinfo/sip=
ping<br>
 &gt; This list is for NEW development of the application of SIP<br>
 &gt; Use sip-implementors@cs.columbia.edu for questions on current sip<br>
 &gt; Use sip@ietf.org for new developments of core SIP<br>
 &gt; <br>
 &gt; <br>
 &gt; <br>
 &gt; <br>
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r>
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From: "Roberto Tealdi" <roby.tea@tin.it>
To: <sip@ietf.org>, <sipping@ietf.org>
Date: Thu, 22 May 2003 15:12:45 +0200
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Subject: [Sipping] [SIP] Hierarchical Registration
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Hello All,

I'm new on SIP, I have a question about registration procedures, =
expecially the Hierarchical Registration described in Shulzrinne's =
article "Application-Layer Mobility Using SIP".

Suppose this scenario, there are 2 domains (Visited Network and Home =
Network) and there is a SIP registrar proxy in each domain;=20

1) a user in the visited network wants to register himself  at his home =
network. UAC sends a REGISTER message; in this message the Contact =
header is set with the current IP address of the UAC. The message has a =
digital signature in order to provide integrity and authentication, such =
digital signature include the following headers: To, From, Contact, =
Date, C-Seq, Call-ID.

2) Suppose to implement the hierarchical registration as described in =
some Schulzrinne's articles: the SIP outbound proxy in the visited =
domain when receives this message has to change the Contact header =
putting in it its own address, before forwarding the REGISTER message to =
the user's home network. This is done in order to receive and inspect =
(by the proxy in the visited network) all incoming messages for the user =
visiting in. But if the Contact header is changed, the digital signature =
is compromised.

The UAC should know the "correct" Contact value before doing the digital =
signature in his REGISTER messages.

How can the proxy do to "inform" the UAC about this "correct" value? Are =
there existing mechanism? Is possible to do it with 4xx messages?

Note: the UAC signs his messages with his private key, the visited proxy =
do not know this key.

I was supposing that this would be a possible solution (see the =
Scenario.txt attach):

The Visited Proxy in the "4xx Contact Not Acceptable" notify the UAC the =
correct value to put in the Contact header, and in the successive =
REGISTER message the UAC fill the header with this new value. The =
Visited Register does not need to made any changes in the REGISTER =
message so the digital signature is still valid.

It would be possible this solution or there are other mechanisms?

Thank you very much for Your helps.
Roby.


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Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2600.0" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>
<DIV><FONT face=3DArial size=3D2>Hello All,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I'm new on SIP, I have a question about =

registration procedures, expecially the Hierarchical Registration =
described in=20
Shulzrinne's article "<FONT face=3D"Times New Roman" size=3D4><FONT =
face=3DArial=20
size=3D2>Application-Layer Mobility Using =
SIP".</FONT></FONT></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Suppose this scenario, there are 2 =
domains (Visited=20
Network and Home Network) and there is a SIP registrar proxy in each =
domain;=20
<BR><BR>1) a user in&nbsp;the visited network wants to register =
himself&nbsp; at=20
his home network. UAC sends a REGISTER message; in this message the =
Contact=20
header is set with the current IP address of the UAC. The message has a =
digital=20
signature in order to provide integrity and authentication,&nbsp;such =
digital=20
signature include the following headers: To, From, Contact, Date, C-Seq, =

Call-ID.<BR><BR>2) Suppose to implement the hierarchical registration as =

described in&nbsp;some Schulzrinne's articles: the SIP outbound proxy in =
the=20
visited domain when receives this message has to change the Contact =
header=20
putting in it&nbsp;its own address, before forwarding the REGISTER =
message to=20
the user's home network.&nbsp;This is&nbsp;done in order to receive and =
inspect=20
(by the proxy in the visited network) all incoming messages&nbsp;for the =
user=20
visiting in. But if the Contact header is changed, the digital signature =
is=20
compromised.<BR><BR>The UAC should know the "correct" Contact value =
before doing=20
the digital signature in his REGISTER messages.<BR><BR>How can the proxy =
do to=20
"inform" the UAC about this "correct" value? Are there existing =
mechanism? Is=20
possible to do it with 4xx messages?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Note: the UAC signs his messages with =
his private=20
key, the visited proxy do not know this key.<BR><BR>I was supposing that =
this=20
would be a possible solution (see the Scenario.txt attach):<BR><BR>The =
Visited=20
Proxy in the "4xx Contact Not Acceptable" notify&nbsp;the UAC the =
correct value=20
to put in the Contact header, and in the successive REGISTER message the =
UAC=20
fill the header with this new value. The Visited Register does not need =
to made=20
any changes in the REGISTER message so the digital signature is still=20
valid.</FONT></DIV><FONT face=3DArial size=3D2></FONT></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><FONT face=3DArial =
size=3D2></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><FONT face=3DArial size=3D2>It would be =
possible this=20
solution or there are other mechanisms?</DIV>
<DIV><BR>Thank you very much for Your=20
helps.<BR>Roby.<BR></DIV></FONT></FONT></BODY></HTML>

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------=_NextPart_000_0021_01C32074.99469220
Content-Type: text/plain;
	name="Scenario.txt"
Content-Transfer-Encoding: quoted-printable
Content-Disposition: attachment;
	filename="Scenario.txt"
Content-Transfer-Encoding: quoted-printable
Content-Transfer-Encoding: quoted-printable
Content-Transfer-Encoding: quoted-printable

UAC                                        Visited Registrar             =
               Home Registrar
   |               REGISTER                       |                      =
                    |
   |--------------------------------------------->|                      =
                    |
   |                                              |                      =
                    |
   |      4xx Contact Address Not Acceptable      |                      =
                    |
   |<---------------------------------------------|					     |    =20
   |                                              |                      =
                    |
   |     REGISTER (with new Contact)              |                      =
                    |
   |--------------------------------------------->|                      =
                    |
   |                                              |                  =
REGISTER                |
   |                                              =
|----------------------------------------->|
   |                                              |                      =
                    |
   |                                              |                200 =
OK                    |
   |                                              =
|<-----------------------------------------|
   |                200 OK                        |                      =
                    |
   |<---------------------------------------------|                      =
                    |
   |                                              |                      =
                    |

------=_NextPart_000_0021_01C32074.99469220--


_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From mailnull@www1.ietf.org  Thu May 22 12:48:22 2003
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Message-ID: <53F74F5A7B94D511841C00B0D0AB16F8B8E55C@baker.datcon.co.uk>
From: Nicole Johnson <NJ2@dataconnection.com>
To: "'Rohan Mahy'" <rohan@cisco.com>
Cc: "'sipping@ietf.org'" <sipping@ietf.org>
Subject: RE: [Sipping] [Sipping]Questions on draft-ietf-sipping-connect-re
	use-reqs-00
Date: Thu, 22 May 2003 17:43:46 +0100
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Rohan,

I think that the requirements should be modified to clarify the scope of the
alias parameter.  I suggest we could add "The mechanism SHOULD allow the
reuse of an existing connection for requests and responses sent both outside
a dialog and within one or more dialogs" to the end of requirement 1 in
section 3.

I also have a question about the lifetime of the alias.  Should the alias be
destroyed as soon as the connection is lost, or is there a reason why you
might want to maintain the alias for any time after?  Is it worth clarifying
this in the draft, perhaps at the beginning of section 4 in the overview of
the mechanism?

Thanks,
Nicole  

-----Original Message-----
From: Rohan Mahy [mailto:rohan@cisco.com]
Sent: Tuesday, May 20, 2003 9:09 PM
To: Nicole Johnson
Cc: 'sipping@ietf.org'
Subject: Re: [Sipping] [Sipping]Questions on
draft-ietf-sipping-connect-reuse-reqs-00


Hi,

On Thursday, May 15, 2003, at 07:23 AM, Nicole Johnson wrote:

> I have a couple of questions about 
> draft-ietf-sipping-connect-reuse-reqs-00.

> 1.  Intended scope of alias parameter.
>
> Is the aim of this draft to enable connections to be reused within a 
> single
> dialog, or across multiple dialogs, i.e. can an alias learned within 
> the
> scope of one dialog be used in another?  Is the alias parameter valid 
> on
> out-of-dialog requests?

across multiple dialogs.  it would work on out-of-dialog requests (ex: 
OPTIONS).
perhaps these should be explicit requirements?

> 2.  Target address fails to authenticate.
>
> If a server receives a request containing the alias parameter and the 
> target
> address of the request fails authentication, how should the server 
> react?

it should not create the alias.

> It could drop the message, respond to the target address with a "not
> authorized" response, or continue processing the request as if the 
> "alias"
> parameter was not present.  Is there a recommended behaviour?

the behavior for handling the request should then do whatever normally 
happens when a request without an alias parameter arrives with failed 
or missing authentication.  frequently this is fine and will result in 
a digest authentication challenge (401 or 407).

hope this makes sense.

thanks,
-rohan

> Thanks,
> Nicole
>
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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To: <aki.niemi@nokia.com>
Cc: rohan@cisco.com, sipping@ietf.org, sipping-admin@ietf.org,
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Subject: RE: [Sipping] Push-To-Talk with SIP - Agreed - {End of Thread }
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--
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11717 Exploration Lane, Germantown MD 20876
(O): 301 212 7860 




<aki.niemi@nokia.com>
05/22/2003 10:50 AM

 
        To:     <aroychow@hns.com>
        cc:     <rohan@cisco.com>, <sipping@ietf.org>, <sipping-admin@ietf.org>, 
<thomas.lee@polypix.com>
        Subject:        RE: [Sipping] Push-To-Talk with SIP


Arjun,

Judging from your answer (especially the 'voice over IM' part) I think we 
might have been talking about two different services under the same name. 
Likewise, the fact that in this thread already there have been at least 
three different interpretations of what a push to talk service means for 
SIP, makes me wonder whether this is a worthwhile debate to have in the 
IETF at all. I.e., let the market do the branding game. ;-)

We should come back when there is a specific SIP issue to address 
(whatever the service), IMHO.

Cheers,
Aki


-----Original Message-----
From: ext aroychow@hss.hns.com [mailto:aroychow@hss.hns.com]
Sent: 22 May, 2003 15:18
To: Niemi Aki (NMP/Helsinki)
Cc: aroychow@hns.com; rohan@cisco.com; sipping@ietf.org; 
sipping-admin@ietf.org; thomas.lee@polypix.com
Subject: RE: [Sipping] Push-To-Talk with SIP



Aki, 
By 'session less' I mean not having an INVITE-OK-ACK to actually establish 
the call before you could talk. Application state would be maintained 
during one 'push-release' operation and forgotten after that (except for 
the administrative states like which was the gruop set I was talking to). 

Different algorithms have been suggested for this 'voice over IM' => some 
are also patented @ the uspto (I think someone had posted the patent no. 
in the impp archive some time ago) 

Anyway, therefore, I realize this may not be something that the ietf sip 
grp wants to take up unless there is a compelling reason 
go into into this 2 way radio service. With respect to the original Q, I 
agree that one could conceptualize that a call in session with RTP 
bursts and conferencing thrown in could look like a '2 way radio service' 
= but due to the nature of its persistance of state for a much longer 
time than currently known implementations not using SIP, the charging 
model and complexities would likely not be the same, for what ever its 
worth. 

regds 
arjun 

--
Arjun Roychowdhury @ Hughes Software Systems
11717 Exploration Lane, Germantown MD 20876
(O): 301 212 7860 


aki.niemi@nokia.com 
Sent by: sipping-admin@ietf.org 
05/22/2003 05:32 AM 
 
        To:        <aroychow@hns.com> 
        cc:        <rohan@cisco.com>, <sipping@ietf.org>, 
<sipping-admin@ietf.org>, <thomas.lee@polypix.com> 
        Subject:        RE: [Sipping] Push-To-Talk with SIP



Hi Arjun,

I don't really understand what you mean by 'session less', could you 
elaborate a bit? Do you mean to say there is no application state 
maintained at the endpoints? 

And to me, the original question seemed to be whether there have been any 
drafts or discussions w.r.t. push to talk and SIP in the IETF. And the 
enswer to that would have to be no, but the conferencing work in SIPPING 
seems to be covering related issues, e.g., floor control, and media 
policy.

Regards, 
Aki


> -----Original Message-----
> From: ext Arjun Roychowdhury [mailto:aroychow@hns.com]
> Sent: 21 May, 2003 17:15
> To: Niemi Aki (NMP/Helsinki)
> Cc: rohan@cisco.com; sipping@ietf.org; sipping-admin@ietf.org;
> thomas.lee@polypix.com
> Subject: RE: [Sipping] Push-To-Talk with SIP
> 
> 
> I think to avoid confusion, it may be better to reword this 
> is a 'session 
> less two way radio service',
> with session less being the key item. I believe this was the 
> original 
> question of the poster.
> 
> Whether this particular approach makes sense to be done in SIP and 
> especially in the purview of the IETF  is another question.
> 
> regds
> arjun
> 
> --
> Arjun Roychowdhury @ Hughes Software Systems
> 11717 Exploration Lane, Germantown MD 20876
> (O): 301 212 7860 
> 
> 
> 
> 
> aki.niemi@nokia.com
> Sent by: sipping-admin@ietf.org
> 05/21/2003 08:52 AM
> 
> 
>         To:     <rohan@cisco.com>, <thomas.lee@polypix.com>
>         cc:     <sipping@ietf.org>
>         Subject:        RE: [Sipping] Push-To-Talk with SIP
> 
> 
> Hi Rohan,
> 
> I think this is one way to look at the push to talk feature, 
> but maybe not 
> exactly "mainstream". Most of the push to talk services I'm 
> aware of are 
> similar to a walkie talkie service, where a group of people are 
> participating, out of which only one at a time can send media to the 
> others. "Requesting the floor" in this case is usually done with the 
> "tangent" button.
> 
> So in my opinion, push to talk is a conference, with a 
> specific type of 
> conference and media policy associated with it. As such, I 
> don't think 
> it's the IETF's business to define those policies but 
> provide tools for 
> setting/manipulating them, which I guess the conference DT 
> is already 
> doing. 
> 
>                  -Aki
> 
>  > -----Original Message-----
>  > From: ext Rohan Mahy [mailto:rohan@cisco.com]
>  > Sent: 20 May, 2003 23:15
>  > To: thomas.lee@polypix.com
>  > Cc: sipping@ietf.org
>  > Subject: Re: [Sipping] Push-To-Talk with SIP
>  > 
>  > 
>  > 
>  > Hi,
>  > 
>  > Push to talk can be implemented by merely setting up a 
>  > session between 
>  > the speaker and the target when the UA becomes available 
> and leaving 
>  > the session open, but only sending audio media over that 
>  > session when a 
>  > button is pushed.  The session can be verified for 
> liveness by using 
>  > the session-timer, and the target suddenly reappearing after 
>  > an absence 
>  > can be detected using the registration package.  Perhaps 
>  > someone could 
>  > write a draft describing this usage?
>  > 
>  > thanks,
>  > -rohan
>  > 
>  > 
>  > On Wednesday, May 14, 2003, at 06:46 PM, ???? ??? wrote:
>  > 
>  > > Hello,
>  > > 
>  > > Nokia, Siemens and Ericsson have announced that they are 
>  > developing a 
>  > > push-to-talk application for GSM networks with GPRS and 
>  > for future 3G 
>  > > networks. They want to use SIP for signaling but I 
>  > couldn't find any 
>  > > more detailed information.
>  > > 
>  > > Does someone of you know some detail about the PTT 
>  > signaling with SIP?
>  > > Do they care about arbitration if two people want to talk 
>  > to the same 
>  > > group at the same time? Are there some drafts or 
> discussions about 
>  > > this topics?
>  > > 
>  > > regards,
>  > > thomas lee.
>  > 
>  > _______________________________________________
>  > Sipping mailing list 
> https://www1.ietf.org/mailman/listinfo/sipping
>  > This list is for NEW development of the application of SIP
>  > Use sip-implementors@cs.columbia.edu for questions on current sip
>  > Use sip@ietf.org for new developments of core SIP
>  > 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
> 
> 
> 
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP 
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: RE: [Sipping] Push-To-Talk with SIP
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Arjun,

Judging from your answer (especially the 'voice over IM' part) I think we might have been talking about two different services under the same name. Likewise, the fact that in this thread already there have been at least three different interpretations of what a push to talk service means for SIP, makes me wonder whether this is a worthwhile debate to have in the IETF at all. I.e., let the market do the branding game. ;-)

We should come back when there is a specific SIP issue to address (whatever the service), IMHO.

Cheers,
Aki


-----Original Message-----
From: ext aroychow@hss.hns.com [mailto:aroychow@hss.hns.com]
Sent: 22 May, 2003 15:18
To: Niemi Aki (NMP/Helsinki)
Cc: aroychow@hns.com; rohan@cisco.com; sipping@ietf.org; sipping-admin@ietf.org; thomas.lee@polypix.com
Subject: RE: [Sipping] Push-To-Talk with SIP



Aki, 
By 'session less' I mean not having an INVITE-OK-ACK to actually establish the call before you could talk. Application state would be maintained 
during one 'push-release' operation and forgotten after that (except for the administrative states like which was the gruop set I was talking to). 

Different algorithms have been suggested for this 'voice over IM' => some are also patented @ the uspto (I think someone had posted the patent no. 
in the impp archive some time ago) 

Anyway, therefore, I realize this may not be something that the ietf sip grp wants to take up unless there is a compelling reason 
go into into this 2 way radio service. With respect to the original Q, I agree that one could conceptualize that a call in session with RTP 
bursts and conferencing thrown in could look like a '2 way radio service' = but due to the nature of its persistance of state for a much longer 
time than currently known implementations not using SIP, the charging model and complexities would likely not be the same, for what ever its worth. 

regds 
arjun 

--
Arjun Roychowdhury @ Hughes Software Systems
11717 Exploration Lane, Germantown MD 20876
(O): 301 212 7860   


aki.niemi@nokia.com 
Sent by: sipping-admin@ietf.org 
05/22/2003 05:32 AM 
        
        To:        <aroychow@hns.com> 
        cc:        <rohan@cisco.com>, <sipping@ietf.org>, <sipping-admin@ietf.org>, <thomas.lee@polypix.com> 
        Subject:        RE: [Sipping] Push-To-Talk with SIP



Hi Arjun,

I don't really understand what you mean by 'session less', could you elaborate a bit? Do you mean to say there is no application state maintained at the endpoints? 

And to me, the original question seemed to be whether there have been any drafts or discussions w.r.t. push to talk and SIP in the IETF. And the enswer to that would have to be no, but the conferencing work in SIPPING seems to be covering related issues, e.g., floor control, and media policy.

Regards,                 
Aki


> -----Original Message-----
> From: ext Arjun Roychowdhury [mailto:aroychow@hns.com]
> Sent: 21 May, 2003 17:15
> To: Niemi Aki (NMP/Helsinki)
> Cc: rohan@cisco.com; sipping@ietf.org; sipping-admin@ietf.org;
> thomas.lee@polypix.com
> Subject: RE: [Sipping] Push-To-Talk with SIP
> 
> 
> I think to avoid confusion, it may be better to reword this 
> is a 'session 
> less two way radio service',
> with session less being the key item. I believe this was the 
> original 
> question of the poster.
> 
> Whether this particular approach makes sense to be done in SIP and 
> especially in the purview of the IETF  is another question.
> 
> regds
> arjun
> 
> --
> Arjun Roychowdhury @ Hughes Software Systems
> 11717 Exploration Lane, Germantown MD 20876
> (O): 301 212 7860 
> 
> 
> 
> 
> aki.niemi@nokia.com
> Sent by: sipping-admin@ietf.org
> 05/21/2003 08:52 AM
> 
>  
>         To:     <rohan@cisco.com>, <thomas.lee@polypix.com>
>         cc:     <sipping@ietf.org>
>         Subject:        RE: [Sipping] Push-To-Talk with SIP
> 
> 
> Hi Rohan,
> 
> I think this is one way to look at the push to talk feature, 
> but maybe not 
> exactly "mainstream". Most of the push to talk services I'm 
> aware of are 
> similar to a walkie talkie service, where a group of people are 
> participating, out of which only one at a time can send media to the 
> others. "Requesting the floor" in this case is usually done with the 
> "tangent" button.
> 
> So in my opinion, push to talk is a conference, with a 
> specific type of 
> conference and media policy associated with it. As such, I 
> don't think 
> it's the IETF's business to define those policies but 
> provide tools for 
> setting/manipulating them, which I guess the conference DT 
> is already 
> doing. 
> 
>                  -Aki
> 
>  > -----Original Message-----
>  > From: ext Rohan Mahy [mailto:rohan@cisco.com]
>  > Sent: 20 May, 2003 23:15
>  > To: thomas.lee@polypix.com
>  > Cc: sipping@ietf.org
>  > Subject: Re: [Sipping] Push-To-Talk with SIP
>  > 
>  > 
>  > 
>  > Hi,
>  > 
>  > Push to talk can be implemented by merely setting up a 
>  > session between 
>  > the speaker and the target when the UA becomes available 
> and leaving 
>  > the session open, but only sending audio media over that 
>  > session when a 
>  > button is pushed.  The session can be verified for 
> liveness by using 
>  > the session-timer, and the target suddenly reappearing after 
>  > an absence 
>  > can be detected using the registration package.  Perhaps 
>  > someone could 
>  > write a draft describing this usage?
>  > 
>  > thanks,
>  > -rohan
>  > 
>  > 
>  > On Wednesday, May 14, 2003, at 06:46 PM, ???? ??? wrote:
>  > 
>  > > Hello,
>  > >  
>  > > Nokia, Siemens and Ericsson have announced that they are 
>  > developing a 
>  > > push-to-talk application for GSM networks with GPRS and 
>  > for future 3G 
>  > > networks. They want to use SIP for signaling but I 
>  > couldn't find any 
>  > > more detailed information.
>  > >  
>  > > Does someone of you know some detail about the PTT 
>  > signaling with SIP?
>  > > Do they care about arbitration if two people want to talk 
>  > to the same 
>  > > group at the same time? Are there some drafts or 
> discussions about 
>  > > this topics?
>  > >  
>  > > regards,
>  > > thomas lee.
>  > 
>  > _______________________________________________
>  > Sipping mailing list  
> https://www1.ietf.org/mailman/listinfo/sipping
>  > This list is for NEW development of the application of SIP
>  > Use sip-implementors@cs.columbia.edu for questions on current sip
>  > Use sip@ietf.org for new developments of core SIP
>  > 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
> 
> 
> 
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP 
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From mailnull@www1.ietf.org  Fri May 23 13:55:41 2003
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To: <Markus.Isomaki@nokia.com>
Cc: aki.niemi@nokia.com, rohan@cisco.com, sipping@ietf.org,
        sipping-admin@ietf.org, thomas.lee@polypix.com
Subject: RE: [Sipping] Push-To-Talk with SIP
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Markus, I agree.
Which is why I referred to the service as '2 way radio' and not PTT , 
since that term is somewhat synonymous to a particular
implementation methodology.

Aki and I discussed this in offline e-mails and concluded for this very 
reason that it doesnt make sense to continue
this thread , at least for now, in the IETF :-)

regds
arjun

--
Arjun Roychowdhury @ Hughes Software Systems
11717 Exploration Lane, Germantown MD 20876
(O): 301 212 7860 




<Markus.Isomaki@nokia.com>
05/23/2003 09:20 AM

 
        To:     aroychow@hns.com, <aki.niemi@nokia.com>
        cc:     <aroychow@hns.com>, <rohan@cisco.com>, <sipping@ietf.org>, 
<sipping-admin@ietf.org>, <thomas.lee@polypix.com>
        Subject:        RE: [Sipping] Push-To-Talk with SIP


Hi Arjun,
 
I guess there are several ways to implement "push to talk" type of 
service. It depends what kind of requirements you have. Sending voice 
clips inside IM is one type of service, but if you want to do multi-party 
communication and floor control, that may not be the best way. Another way 
is to model the communication as conferences (as in SIPPING Conferencing 
Framework), in which case you will use INVITE-OK-ACK to join and BYE to 
leave. But as said, there are many ways to implement the actual service 
using SIP, SIMPLE, conferencing specifications, floor control etc. that it 
does not seem something that really fits into IETF scope. Of course IETF 
should continue developing the basic building blocks such as messaging and 
conferencing related specifications.
 
Markus
-----Original Message-----
From: ext aroychow@hss.hns.com [mailto:aroychow@hss.hns.com]
Sent: 22 May, 2003 15:18
To: Niemi Aki (NMP/Helsinki)
Cc: aroychow@hns.com; rohan@cisco.com; sipping@ietf.org; 
sipping-admin@ietf.org; thomas.lee@polypix.com
Subject: RE: [Sipping] Push-To-Talk with SIP


Aki, 
By 'session less' I mean not having an INVITE-OK-ACK to actually establish 
the call before you could talk. Application state would be maintained 
during one 'push-release' operation and forgotten after that (except for 
the administrative states like which was the gruop set I was talking to). 

Different algorithms have been suggested for this 'voice over IM' => some 
are also patented @ the uspto (I think someone had posted the patent no. 
in the impp archive some time ago) 

Anyway, therefore, I realize this may not be something that the ietf sip 
grp wants to take up unless there is a compelling reason 
go into into this 2 way radio service. With respect to the original Q, I 
agree that one could conceptualize that a call in session with RTP 
bursts and conferencing thrown in could look like a '2 way radio service' 
= but due to the nature of its persistance of state for a much longer 
time than currently known implementations not using SIP, the charging 
model and complexities would likely not be the same, for what ever its 
worth. 

regds 
arjun 

--
Arjun Roychowdhury @ Hughes Software Systems
11717 Exploration Lane, Germantown MD 20876
(O): 301 212 7860   



aki.niemi@nokia.com 
Sent by: sipping-admin@ietf.org 
05/22/2003 05:32 AM 
        
        To:        <aroychow@hns.com> 
        cc:        <rohan@cisco.com>, <sipping@ietf.org>, 
<sipping-admin@ietf.org>, <thomas.lee@polypix.com> 
        Subject:        RE: [Sipping] Push-To-Talk with SIP



Hi Arjun,

I don't really understand what you mean by 'session less', could you 
elaborate a bit? Do you mean to say there is no application state 
maintained at the endpoints? 

And to me, the original question seemed to be whether there have been any 
drafts or discussions w.r.t. push to talk and SIP in the IETF. And the 
enswer to that would have to be no, but the conferencing work in SIPPING 
seems to be covering related issues, e.g., floor control, and media 
policy.

Regards, 
Aki


> -----Original Message-----
> From: ext Arjun Roychowdhury [mailto:aroychow@hns.com]
> Sent: 21 May, 2003 17:15
> To: Niemi Aki (NMP/Helsinki)
> Cc: rohan@cisco.com; sipping@ietf.org; sipping-admin@ietf.org;
> thomas.lee@polypix.com
> Subject: RE: [Sipping] Push-To-Talk with SIP
> 
> 
> I think to avoid confusion, it may be better to reword this 
> is a 'session 
> less two way radio service',
> with session less being the key item. I believe this was the 
> original 
> question of the poster.
> 
> Whether this particular approach makes sense to be done in SIP and 
> especially in the purview of the IETF  is another question.
> 
> regds
> arjun
> 
> --
> Arjun Roychowdhury @ Hughes Software Systems
> 11717 Exploration Lane, Germantown MD 20876
> (O): 301 212 7860 
> 
> 
> 
> 
> aki.niemi@nokia.com
> Sent by: sipping-admin@ietf.org
> 05/21/2003 08:52 AM
> 
> 
>         To:     <rohan@cisco.com>, <thomas.lee@polypix.com>
>         cc:     <sipping@ietf.org>
>         Subject:        RE: [Sipping] Push-To-Talk with SIP
> 
> 
> Hi Rohan,
> 
> I think this is one way to look at the push to talk feature, 
> but maybe not 
> exactly "mainstream". Most of the push to talk services I'm 
> aware of are 
> similar to a walkie talkie service, where a group of people are 
> participating, out of which only one at a time can send media to the 
> others. "Requesting the floor" in this case is usually done with the 
> "tangent" button.
> 
> So in my opinion, push to talk is a conference, with a 
> specific type of 
> conference and media policy associated with it. As such, I 
> don't think 
> it's the IETF's business to define those policies but 
> provide tools for 
> setting/manipulating them, which I guess the conference DT 
> is already 
> doing. 
> 
>                  -Aki
> 
>  > -----Original Message-----
>  > From: ext Rohan Mahy [mailto:rohan@cisco.com]
>  > Sent: 20 May, 2003 23:15
>  > To: thomas.lee@polypix.com
>  > Cc: sipping@ietf.org
>  > Subject: Re: [Sipping] Push-To-Talk with SIP
>  > 
>  > 
>  > 
>  > Hi,
>  > 
>  > Push to talk can be implemented by merely setting up a 
>  > session between 
>  > the speaker and the target when the UA becomes available 
> and leaving 
>  > the session open, but only sending audio media over that 
>  > session when a 
>  > button is pushed.  The session can be verified for 
> liveness by using 
>  > the session-timer, and the target suddenly reappearing after 
>  > an absence 
>  > can be detected using the registration package.  Perhaps 
>  > someone could 
>  > write a draft describing this usage?
>  > 
>  > thanks,
>  > -rohan
>  > 
>  > 
>  > On Wednesday, May 14, 2003, at 06:46 PM, ???? ??? wrote:
>  > 
>  > > Hello,
>  > > 
>  > > Nokia, Siemens and Ericsson have announced that they are 
>  > developing a 
>  > > push-to-talk application for GSM networks with GPRS and 
>  > for future 3G 
>  > > networks. They want to use SIP for signaling but I 
>  > couldn't find any 
>  > > more detailed information.
>  > > 
>  > > Does someone of you know some detail about the PTT 
>  > signaling with SIP?
>  > > Do they care about arbitration if two people want to talk 
>  > to the same 
>  > > group at the same time? Are there some drafts or 
> discussions about 
>  > > this topics?
>  > > 
>  > > regards,
>  > > thomas lee.
>  > 
>  > _______________________________________________
>  > Sipping mailing list 
> https://www1.ietf.org/mailman/listinfo/sipping
>  > This list is for NEW development of the application of SIP
>  > Use sip-implementors@cs.columbia.edu for questions on current sip
>  > Use sip@ietf.org for new developments of core SIP
>  > 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
> 
> 
> 
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP 
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP




_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: RE: [Sipping] Push-To-Talk with SIP
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Hi Arjun,
=20
I guess there are several ways to implement "push to talk" type of =
service. It depends what kind of requirements you have. Sending voice =
clips inside IM is one type of service, but if you want to do =
multi-party communication and floor control, that may not be the best =
way. Another way is to model the communication as conferences (as in =
SIPPING Conferencing Framework), in which case you will use =
INVITE-OK-ACK to join and BYE to leave. But as said, there are many ways =
to implement the actual service using SIP, SIMPLE, conferencing =
specifications, floor control etc. that it does not seem something that =
really fits into IETF scope. Of course IETF should continue developing =
the basic building blocks such as messaging and conferencing related =
specifications.
=20
Markus

-----Original Message-----
From: ext aroychow@hss.hns.com [mailto:aroychow@hss.hns.com]
Sent: 22 May, 2003 15:18
To: Niemi Aki (NMP/Helsinki)
Cc: aroychow@hns.com; rohan@cisco.com; sipping@ietf.org; =
sipping-admin@ietf.org; thomas.lee@polypix.com
Subject: RE: [Sipping] Push-To-Talk with SIP



Aki,=20
By 'session less' I mean not having an INVITE-OK-ACK to actually =
establish the call before you could talk. Application state would be =
maintained=20
during one 'push-release' operation and forgotten after that (except for =
the administrative states like which was the gruop set I was talking =
to).=20

Different algorithms have been suggested for this 'voice over IM' =3D> =
some are also patented @ the uspto (I think someone had posted the =
patent no.=20
in the impp archive some time ago)=20

Anyway, therefore, I realize this may not be something that the ietf sip =
grp wants to take up unless there is a compelling reason=20
go into into this 2 way radio service. With respect to the original Q, I =
agree that one could conceptualize that a call in session with RTP=20
bursts and conferencing thrown in could look like a '2 way radio =
service' =3D but due to the nature of its persistance of state for a =
much longer=20
time than currently known implementations not using SIP, the charging =
model and complexities would likely not be the same, for what ever its =
worth.=20

regds=20
arjun=20

--
Arjun Roychowdhury @ Hughes Software Systems
11717 Exploration Lane, Germantown MD 20876
(O): 301 212 7860  =20



	aki.niemi@nokia.com=20
Sent by: sipping-admin@ietf.org=20


05/22/2003 05:32 AM=20


       =20
        To:        <aroychow@hns.com>=20
        cc:        <rohan@cisco.com>, <sipping@ietf.org>, =
<sipping-admin@ietf.org>, <thomas.lee@polypix.com>=20
        Subject:        RE: [Sipping] Push-To-Talk with SIP



Hi Arjun,

I don't really understand what you mean by 'session less', could you =
elaborate a bit? Do you mean to say there is no application state =
maintained at the endpoints?=20

And to me, the original question seemed to be whether there have been =
any drafts or discussions w.r.t. push to talk and SIP in the IETF. And =
the enswer to that would have to be no, but the conferencing work in =
SIPPING seems to be covering related issues, e.g., floor control, and =
media policy.

Regards,                =20
Aki


> -----Original Message-----
> From: ext Arjun Roychowdhury [mailto:aroychow@hns.com]
> Sent: 21 May, 2003 17:15
> To: Niemi Aki (NMP/Helsinki)
> Cc: rohan@cisco.com; sipping@ietf.org; sipping-admin@ietf.org;
> thomas.lee@polypix.com
> Subject: RE: [Sipping] Push-To-Talk with SIP
>=20
>=20
> I think to avoid confusion, it may be better to reword this=20
> is a 'session=20
> less two way radio service',
> with session less being the key item. I believe this was the=20
> original=20
> question of the poster.
>=20
> Whether this particular approach makes sense to be done in SIP and=20
> especially in the purview of the IETF  is another question.
>=20
> regds
> arjun
>=20
> --
> Arjun Roychowdhury @ Hughes Software Systems
> 11717 Exploration Lane, Germantown MD 20876
> (O): 301 212 7860=20
>=20
>=20
>=20
>=20
> aki.niemi@nokia.com
> Sent by: sipping-admin@ietf.org
> 05/21/2003 08:52 AM
>=20
> =20
>         To:     <rohan@cisco.com>, <thomas.lee@polypix.com>
>         cc:     <sipping@ietf.org>
>         Subject:        RE: [Sipping] Push-To-Talk with SIP
>=20
>=20
> Hi Rohan,
>=20
> I think this is one way to look at the push to talk feature,=20
> but maybe not=20
> exactly "mainstream". Most of the push to talk services I'm=20
> aware of are=20
> similar to a walkie talkie service, where a group of people are=20
> participating, out of which only one at a time can send media to the=20
> others. "Requesting the floor" in this case is usually done with the=20
> "tangent" button.
>=20
> So in my opinion, push to talk is a conference, with a=20
> specific type of=20
> conference and media policy associated with it. As such, I=20
> don't think=20
> it's the IETF's business to define those policies but=20
> provide tools for=20
> setting/manipulating them, which I guess the conference DT=20
> is already=20
> doing.=20
>=20
>                  -Aki
>=20
>  > -----Original Message-----
>  > From: ext Rohan Mahy [mailto:rohan@cisco.com]
>  > Sent: 20 May, 2003 23:15
>  > To: thomas.lee@polypix.com
>  > Cc: sipping@ietf.org
>  > Subject: Re: [Sipping] Push-To-Talk with SIP
>  >=20
>  >=20
>  >=20
>  > Hi,
>  >=20
>  > Push to talk can be implemented by merely setting up a=20
>  > session between=20
>  > the speaker and the target when the UA becomes available=20
> and leaving=20
>  > the session open, but only sending audio media over that=20
>  > session when a=20
>  > button is pushed.  The session can be verified for=20
> liveness by using=20
>  > the session-timer, and the target suddenly reappearing after=20
>  > an absence=20
>  > can be detected using the registration package.  Perhaps=20
>  > someone could=20
>  > write a draft describing this usage?
>  >=20
>  > thanks,
>  > -rohan
>  >=20
>  >=20
>  > On Wednesday, May 14, 2003, at 06:46 PM, ???? ??? wrote:
>  >=20
>  > > Hello,
>  > > =20
>  > > Nokia, Siemens and Ericsson have announced that they are=20
>  > developing a=20
>  > > push-to-talk application for GSM networks with GPRS and=20
>  > for future 3G=20
>  > > networks. They want to use SIP for signaling but I=20
>  > couldn't find any=20
>  > > more detailed information.
>  > > =20
>  > > Does someone of you know some detail about the PTT=20
>  > signaling with SIP?
>  > > Do they care about arbitration if two people want to talk=20
>  > to the same=20
>  > > group at the same time? Are there some drafts or=20
> discussions about=20
>  > > this topics?
>  > > =20
>  > > regards,
>  > > thomas lee.
>  >=20
>  > _______________________________________________
>  > Sipping mailing list =20
> https://www1.ietf.org/mailman/listinfo/sipping
>  > This list is for NEW development of the application of SIP
>  > Use sip-implementors@cs.columbia.edu for questions on current sip
>  > Use sip@ietf.org for new developments of core SIP
>  >=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20
>=20
>=20
>=20
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP=20
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP





------_=_NextPart_001_01C3212E.1DF434B6
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">


<META content=3D"MSHTML 5.50.4611.1300" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D264061313-23052003><FONT face=3DArial color=3D#0000ff =
size=3D2>Hi=20
Arjun,</FONT></SPAN></DIV>
<DIV><SPAN class=3D264061313-23052003><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D264061313-23052003><FONT face=3DArial color=3D#0000ff =
size=3D2>I=20
guess there are several ways to implement "push to talk" type of =
service. It=20
depends what kind of requirements you have. Sending voice clips inside =
IM is one=20
type of service, but if you want to do multi-party communication and =
floor=20
control, that may not be the best way. Another way is to&nbsp;model the=20
communication&nbsp;as conferences (as in SIPPING Conferencing =
Framework), in=20
which case you will use INVITE-OK-ACK to join and BYE to leave. But as =
said,=20
there are many ways to implement the actual service using SIP, SIMPLE,=20
conferencing specifications, floor control etc. that it does not seem =
something=20
that really fits into IETF scope. Of course IETF should continue =
developing the=20
basic building blocks such as messaging and conferencing related=20
specifications.</FONT></SPAN></DIV>
<DIV><SPAN class=3D264061313-23052003><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D264061313-23052003><FONT face=3DArial color=3D#0000ff =

size=3D2>Markus</FONT></SPAN></DIV>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> ext =
aroychow@hss.hns.com=20
  [mailto:aroychow@hss.hns.com]<BR><B>Sent:</B> 22 May, 2003 =
15:18<BR><B>To:</B>=20
  Niemi Aki (NMP/Helsinki)<BR><B>Cc:</B> aroychow@hns.com; =
rohan@cisco.com;=20
  sipping@ietf.org; sipping-admin@ietf.org;=20
  thomas.lee@polypix.com<BR><B>Subject:</B> RE: [Sipping] Push-To-Talk =
with=20
  SIP<BR><BR></FONT></DIV><BR><FONT face=3Dsans-serif =
size=3D2>Aki,</FONT> <BR><FONT=20
  face=3Dsans-serif size=3D2>By 'session less' I mean not having an =
INVITE-OK-ACK to=20
  actually establish the call before you could talk. Application state =
would be=20
  maintained</FONT> <BR><FONT face=3Dsans-serif size=3D2>during one =
'push-release'=20
  operation and forgotten after that (except for the administrative =
states like=20
  which was the gruop set I was talking to).</FONT> <BR><BR><FONT=20
  face=3Dsans-serif size=3D2>Different algorithms have been suggested =
for this=20
  'voice over IM' =3D&gt; some are also patented @ the uspto (I think =
someone had=20
  posted the patent no.</FONT> <BR><FONT face=3Dsans-serif size=3D2>in =
the impp=20
  archive some time ago)</FONT> <BR><BR><FONT face=3Dsans-serif =
size=3D2>Anyway,=20
  therefore, I realize this may not be something that the ietf sip grp =
wants to=20
  take up unless there is a compelling reason </FONT><BR><FONT =
face=3Dsans-serif=20
  size=3D2>go into into this 2 way radio service. With respect to the =
original Q,=20
  I agree that one could conceptualize that a call in session with =
RTP</FONT>=20
  <BR><FONT face=3Dsans-serif size=3D2>bursts and conferencing thrown in =
could look=20
  like a '2 way radio service' =3D but due to the nature of its =
persistance of=20
  state for a much longer</FONT> <BR><FONT face=3Dsans-serif =
size=3D2>time than=20
  currently known implementations not using SIP, the charging model and=20
  complexities would likely not be the same, for what ever its =
worth.</FONT>=20
  <BR><BR><FONT face=3Dsans-serif size=3D2>regds</FONT> <BR><FONT =
face=3Dsans-serif=20
  size=3D2>arjun</FONT> <BR><BR><FONT face=3Dsans-serif =
size=3D2>--<BR>Arjun=20
  Roychowdhury @ Hughes Software Systems<BR>11717 Exploration Lane, =
Germantown=20
  MD 20876<BR>(O): 301 212 7860 &nbsp;</FONT> <BR><BR><BR>
  <TABLE width=3D"100%">
    <TBODY>
    <TR vAlign=3Dtop>
      <TD>
      <TD><FONT face=3Dsans-serif =
size=3D1><B>aki.niemi@nokia.com</B></FONT>=20
        <BR><FONT face=3Dsans-serif size=3D1>Sent by: =
sipping-admin@ietf.org</FONT>=20
        <P><FONT face=3Dsans-serif size=3D1>05/22/2003 05:32 AM</FONT> =
<BR></P>
      <TD><FONT face=3DArial size=3D1>&nbsp; &nbsp; &nbsp; &nbsp; =
</FONT><BR><FONT=20
        face=3Dsans-serif size=3D1>&nbsp; &nbsp; &nbsp; &nbsp; To: =
&nbsp; &nbsp;=20
        &nbsp; &nbsp;&lt;aroychow@hns.com&gt;</FONT> <BR><FONT =
face=3Dsans-serif=20
        size=3D1>&nbsp; &nbsp; &nbsp; &nbsp; cc: &nbsp; &nbsp; &nbsp;=20
        &nbsp;&lt;rohan@cisco.com&gt;, &lt;sipping@ietf.org&gt;,=20
        &lt;sipping-admin@ietf.org&gt;, =
&lt;thomas.lee@polypix.com&gt;</FONT>=20
        <BR><FONT face=3Dsans-serif size=3D1>&nbsp; &nbsp; &nbsp; &nbsp; =
Subject:=20
        &nbsp; &nbsp; &nbsp; &nbsp;RE: [Sipping] Push-To-Talk with=20
    SIP</FONT></TR></TBODY></TABLE><BR><BR><BR><FONT face=3D"Courier =
New" size=3D2>Hi=20
  Arjun,<BR><BR>I don't really understand what you mean by 'session =
less', could=20
  you elaborate a bit? Do you mean to say there is no application state=20
  maintained at the endpoints? <BR><BR>And to me, the original question =
seemed=20
  to be whether there have been any drafts or discussions w.r.t. push to =
talk=20
  and SIP in the IETF. And the enswer to that would have to be no, but =
the=20
  conferencing work in SIPPING seems to be covering related issues, =
e.g., floor=20
  control, and media policy.<BR><BR>Regards, &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
  &nbsp; &nbsp; &nbsp; <BR>Aki<BR><BR><BR>&gt; -----Original=20
  Message-----<BR>&gt; From: ext Arjun Roychowdhury=20
  [mailto:aroychow@hns.com]<BR>&gt; Sent: 21 May, 2003 17:15<BR>&gt; To: =
Niemi=20
  Aki (NMP/Helsinki)<BR>&gt; Cc: rohan@cisco.com; sipping@ietf.org;=20
  sipping-admin@ietf.org;<BR>&gt; thomas.lee@polypix.com<BR>&gt; =
Subject: RE:=20
  [Sipping] Push-To-Talk with SIP<BR>&gt; <BR>&gt; <BR>&gt; I think to =
avoid=20
  confusion, it may be better to reword this <BR>&gt; is a 'session =
<BR>&gt;=20
  less two way radio service',<BR>&gt; with session less being the key =
item. I=20
  believe this was the <BR>&gt; original <BR>&gt; question of the=20
  poster.<BR>&gt; <BR>&gt; Whether this particular approach makes sense =
to be=20
  done in SIP and <BR>&gt; especially in the purview of the IETF =
&nbsp;is=20
  another question.<BR>&gt; <BR>&gt; regds<BR>&gt; arjun<BR>&gt; =
<BR>&gt;=20
  --<BR>&gt; Arjun Roychowdhury @ Hughes Software Systems<BR>&gt; 11717=20
  Exploration Lane, Germantown MD 20876<BR>&gt; (O): 301 212 7860 =
<BR>&gt;=20
  <BR>&gt; <BR>&gt; <BR>&gt; <BR>&gt; aki.niemi@nokia.com<BR>&gt; Sent =
by:=20
  sipping-admin@ietf.org<BR>&gt; 05/21/2003 08:52 AM<BR>&gt; <BR>&gt;=20
  &nbsp;<BR>&gt; &nbsp; &nbsp; &nbsp; &nbsp; To: &nbsp; &nbsp;=20
  &lt;rohan@cisco.com&gt;, &lt;thomas.lee@polypix.com&gt;<BR>&gt; &nbsp; =
&nbsp;=20
  &nbsp; &nbsp; cc: &nbsp; &nbsp; &lt;sipping@ietf.org&gt;<BR>&gt; =
&nbsp; &nbsp;=20
  &nbsp; &nbsp; Subject: &nbsp; &nbsp; &nbsp; &nbsp;RE: [Sipping] =
Push-To-Talk=20
  with SIP<BR>&gt; <BR>&gt; <BR>&gt; Hi Rohan,<BR>&gt; <BR>&gt; I think =
this is=20
  one way to look at the push to talk feature, <BR>&gt; but maybe not =
<BR>&gt;=20
  exactly "mainstream". Most of the push to talk services I'm <BR>&gt; =
aware of=20
  are <BR>&gt; similar to a walkie talkie service, where a group of =
people are=20
  <BR>&gt; participating, out of which only one at a time can send media =
to the=20
  <BR>&gt; others. "Requesting the floor" in this case is usually done =
with the=20
  <BR>&gt; "tangent" button.<BR>&gt; <BR>&gt; So in my opinion, push to =
talk is=20
  a conference, with a <BR>&gt; specific type of <BR>&gt; conference and =
media=20
  policy associated with it. As such, I <BR>&gt; don't think <BR>&gt; =
it's the=20
  IETF's business to define those policies but <BR>&gt; provide tools =
for=20
  <BR>&gt; setting/manipulating them, which I guess the conference DT =
<BR>&gt;=20
  is already <BR>&gt; doing. <BR>&gt; <BR>&gt; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
  &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;-Aki<BR>&gt; <BR>&gt; &nbsp;&gt;=20
  -----Original Message-----<BR>&gt; &nbsp;&gt; From: ext Rohan Mahy=20
  [mailto:rohan@cisco.com]<BR>&gt; &nbsp;&gt; Sent: 20 May, 2003 =
23:15<BR>&gt;=20
  &nbsp;&gt; To: thomas.lee@polypix.com<BR>&gt; &nbsp;&gt; Cc:=20
  sipping@ietf.org<BR>&gt; &nbsp;&gt; Subject: Re: [Sipping] =
Push-To-Talk with=20
  SIP<BR>&gt; &nbsp;&gt; <BR>&gt; &nbsp;&gt; <BR>&gt; &nbsp;&gt; =
<BR>&gt;=20
  &nbsp;&gt; Hi,<BR>&gt; &nbsp;&gt; <BR>&gt; &nbsp;&gt; Push to talk can =
be=20
  implemented by merely setting up a <BR>&gt; &nbsp;&gt; session between =

  <BR>&gt; &nbsp;&gt; the speaker and the target when the UA becomes =
available=20
  <BR>&gt; and leaving <BR>&gt; &nbsp;&gt; the session open, but only =
sending=20
  audio media over that <BR>&gt; &nbsp;&gt; session when a <BR>&gt; =
&nbsp;&gt;=20
  button is pushed. &nbsp;The session can be verified for <BR>&gt; =
liveness by=20
  using <BR>&gt; &nbsp;&gt; the session-timer, and the target suddenly=20
  reappearing after <BR>&gt; &nbsp;&gt; an absence <BR>&gt; &nbsp;&gt; =
can be=20
  detected using the registration package. &nbsp;Perhaps <BR>&gt; =
&nbsp;&gt;=20
  someone could <BR>&gt; &nbsp;&gt; write a draft describing this =
usage?<BR>&gt;=20
  &nbsp;&gt; <BR>&gt; &nbsp;&gt; thanks,<BR>&gt; &nbsp;&gt; =
-rohan<BR>&gt;=20
  &nbsp;&gt; <BR>&gt; &nbsp;&gt; <BR>&gt; &nbsp;&gt; On Wednesday, May =
14, 2003,=20
  at 06:46 PM, ???? ??? wrote:<BR>&gt; &nbsp;&gt; <BR>&gt; &nbsp;&gt; =
&gt;=20
  Hello,<BR>&gt; &nbsp;&gt; &gt; &nbsp;<BR>&gt; &nbsp;&gt; &gt; Nokia, =
Siemens=20
  and Ericsson have announced that they are <BR>&gt; &nbsp;&gt; =
developing a=20
  <BR>&gt; &nbsp;&gt; &gt; push-to-talk application for GSM networks =
with GPRS=20
  and <BR>&gt; &nbsp;&gt; for future 3G <BR>&gt; &nbsp;&gt; &gt; =
networks. They=20
  want to use SIP for signaling but I <BR>&gt; &nbsp;&gt; couldn't find =
any=20
  <BR>&gt; &nbsp;&gt; &gt; more detailed information.<BR>&gt; &nbsp;&gt; =
&gt;=20
  &nbsp;<BR>&gt; &nbsp;&gt; &gt; Does someone of you know some detail =
about the=20
  PTT <BR>&gt; &nbsp;&gt; signaling with SIP?<BR>&gt; &nbsp;&gt; &gt; Do =
they=20
  care about arbitration if two people want to talk <BR>&gt; &nbsp;&gt; =
to the=20
  same <BR>&gt; &nbsp;&gt; &gt; group at the same time? Are there some =
drafts or=20
  <BR>&gt; discussions about <BR>&gt; &nbsp;&gt; &gt; this =
topics?<BR>&gt;=20
  &nbsp;&gt; &gt; &nbsp;<BR>&gt; &nbsp;&gt; &gt; regards,<BR>&gt; =
&nbsp;&gt;=20
  &gt; thomas lee.<BR>&gt; &nbsp;&gt; <BR>&gt; &nbsp;&gt;=20
  _______________________________________________<BR>&gt; &nbsp;&gt; =
Sipping=20
  mailing list &nbsp;<BR>&gt;=20
  https://www1.ietf.org/mailman/listinfo/sipping<BR>&gt; &nbsp;&gt; This =
list is=20
  for NEW development of the application of SIP<BR>&gt; &nbsp;&gt; Use=20
  sip-implementors@cs.columbia.edu for questions on current sip<BR>&gt;=20
  &nbsp;&gt; Use sip@ietf.org for new developments of core SIP<BR>&gt;=20
  &nbsp;&gt; <BR>&gt; =
_______________________________________________<BR>&gt;=20
  Sipping mailing list=20
  &nbsp;https://www1.ietf.org/mailman/listinfo/sipping<BR>&gt; This list =
is for=20
  NEW development of the application of SIP<BR>&gt; Use=20
  sip-implementors@cs.columbia.edu for questions on current sip<BR>&gt; =
Use=20
  sip@ietf.org for new developments of core SIP<BR>&gt; <BR>&gt; =
<BR>&gt;=20
  <BR>&gt; =
<BR>_______________________________________________<BR>Sipping=20
  mailing list =
&nbsp;https://www1.ietf.org/mailman/listinfo/sipping<BR>This list=20
  is for NEW development of the application of SIP</FONT> <BR><FONT=20
  face=3D"Courier New" size=3D2>Use sip-implementors@cs.columbia.edu for =
questions=20
  on current sip<BR>Use sip@ietf.org for new developments of core=20
  SIP<BR></FONT><BR><BR></BLOCKQUOTE></BODY></HTML>

------_=_NextPart_001_01C3212E.1DF434B6--
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From mailnull@www1.ietf.org  Tue May 27 02:07:13 2003
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Message-ID: <F8EFC4B4A8C016428BC1F589296D4FBFBC1139@esealnt630.al.sw.ericsson.se>
From: "Gonzalo Camarillo Gonzalez (LMF)"
	 <Gonzalo.Camarillo@lmf.ericsson.se>
To: "'sipping@ietf.org'" <sipping@ietf.org>
Cc: "Rohan Mahy (E-mail)" <rohan@cisco.com>,
        "Dean Willis (E-mail)"
	 <dean.willis@softarmor.com>,
        "Patrick Mourot (E-mail)"
	 <pm@sxb.bsf.alcatel.fr>,
        "Olivier Rousseau (E-mail)"
	 <olivier.rousseau@col.bsf.alcatel.fr>,
        "Frank Derks (E-mail)"
	 <frank.derks@philips.com>
Date: Tue, 27 May 2003 08:03:55 +0200
MIME-Version: 1.0
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Subject: [Sipping] WGLC for Q.SIP-SIP mapping draft
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Folks,

we would like to WGLC the draft below:

http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-01.txt

This WGLC will end on June 20th.

Gonzalo
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Subject: [Sipping] I-D ACTION:draft-elwell-sipping-identity-interworking-00.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: User identification in a SIP/QSIG environment
	Author(s)	: J. Elwell
	Filename	: draft-elwell-sipping-identity-interworking-00.txt
	Pages		: 31
	Date		: 2003-5-28
	
This document examines means of identifying or naming users of 
telephony services within an enterprise. Numeric names (numbers) are 
used in traditional Private Integrated Services Networks (PISNs) 
using QSIG as the network signalling protocol. They are also used for 
external communication, e.g., with a public Integrated Services 
Digital Network (ISDN). Names need not be numeric in Internet 
Protocol (IP) networks employing signalling protocols such as the 
Session Initiation Protocol (SIP). This document therefore looks at 
naming schemes that are appropriate within enterprise IP networks, in 
particular enterprise IP networks employing SIP as the signalling 
protocol. It also investigates naming schemes that are appropriate in 
a mixed QSIG/SIP enterprise network and the treatment of names at an 
interworking point. It details the use of names not only for 
selecting a user to participate in a call, but also as a means of 
identifying a user in a call to other users in that call. ENUM and 
private ENUM-like services are also examined.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-elwell-sipping-identity-interworking-00.txt

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Date: Thu, 29 May 2003 15:37:38 +0100
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sipping] I-D ACTION:draft-elwell-sipping-identity-interworki
	ng-00.txt
To: "'sipping@ietf.org'" <sipping@ietf.org>
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This document has been produced in ECMA, and comments of the SIP community
are invited prior to finalisation of the ECMA publication during the coming
weeks.

john.elwell@siemens.com


> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> Sent: 29 May 2003 12:20
> Cc: sipping@ietf.org
> Subject: [Sipping] I-D
> ACTION:draft-elwell-sipping-identity-interworking-00.txt
> 
> 
> A New Internet-Draft is available from the on-line 
> Internet-Drafts directories.
> 
> 
> 	Title		: User identification in a SIP/QSIG environment
> 	Author(s)	: J. Elwell
> 	Filename	: 
> draft-elwell-sipping-identity-interworking-00.txt
> 	Pages		: 31
> 	Date		: 2003-5-28
> 	
> This document examines means of identifying or naming users of 
> telephony services within an enterprise. Numeric names (numbers) are 
> used in traditional Private Integrated Services Networks (PISNs) 
> using QSIG as the network signalling protocol. They are also used for 
> external communication, e.g., with a public Integrated Services 
> Digital Network (ISDN). Names need not be numeric in Internet 
> Protocol (IP) networks employing signalling protocols such as the 
> Session Initiation Protocol (SIP). This document therefore looks at 
> naming schemes that are appropriate within enterprise IP networks, in 
> particular enterprise IP networks employing SIP as the signalling 
> protocol. It also investigates naming schemes that are appropriate in 
> a mixed QSIG/SIP enterprise network and the treatment of names at an 
> interworking point. It details the use of names not only for 
> selecting a user to participate in a call, but also as a means of 
> identifying a user in a call to other users in that call. ENUM and 
> private ENUM-like services are also examined.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-elwell-sipping-ident
ity-interworking-00.txt

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From: "Sarkar, Uttam" <Uttam.Sarkar@icn.siemens.com>
To: "'sipping@ietf.org'" <sipping@ietf.org>
Date: Thu, 29 May 2003 15:23:55 -0400
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Subject: [Sipping] SIP and TCP transport....
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Hi All,
I have posted this messgage to the sip-implementors list. There might have
been a discussion already regarding this.
When transport is TCP, do you keep the socket connection open till the call
is active or you close the socket once party A and party B are connected.
When you get BYE or other sip Message from either end, then you open new
socket connection to the other side to send that request. I would appreciate
your inputs.
Thanks,
Uttam Sarkar

             B2BUA
A<----------->|
              |
              |<-------------->B
              |
              |
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Francois Audet" <audet@nortelnetworks.com>
To: "'Elwell, John'" <john.elwell@siemens.com>,
        "'sipping@ietf.org'"
	 <sipping@ietf.org>
Subject: RE: [Sipping] I-D ACTION:draft-elwell-sipping-identity-interworki
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Hi John,

This is a very useful document. 

In any case, I've got a few comments:

- I would like to see a mention of how to relate SIP URIs and tel URIs
  for QSIG phone numbers by putting an explicit reference to section
  19.1.6 of RFC 3261. This could be somewhere in your draft, perhaps in
  section 11.2.2.

- There doesn't seem to be much guidance in your draft of how to 
  indicate QSIG type of number information using the phone-context.
  This is precisely the type of thing that the phone-context was 
  created to do for "local" numbers (i.e., within a private domain).
  For example, a enterprise foo.com might decide to use the following 
  mapping for NPI/TON to phone-context:

  	NPI=Private ISO/IEC 11571, TON=	phone-context=
      -----------------------------------------------------
	level 2 regional number			level2.foo.com
	level 1 regional number			level1.foo.com
	local (level 0) number			level0.foo.com
	PISN-specific number			special.foo.com
	unknown number				unknown.foo.com

  Of course, an enterprise could choose to use a phone-context of foo.com
  as the highest level of the phone number hierarchy, be it Level 2 or Level
1.
  Also, the host could be diffent that the domain in the phone-context: you
  could have foo.com in the phone-context and serviceprovider.net as the
host
  domain. Also, this is just an example: the idea is that the enterprise
  would assign diffent phone contextes to different QSIG types of number.

  This would yield SIP URIs like 
	sip:53756;phone-context=level0.foo.com@foo.com;user=phone
	sip:265-3756;phone-context=level1.foo.com@foo.com;user=phone
	sip:6-911;phone-context=secial.foo.com@foo.com;user=phone
  
  This is probably something that should also go in section 11.2.2.
  
  All this is very much in line with draft-antti-rfc2806bis-08.

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com] 
> Sent: Thursday, May 29, 2003 07:38
> To: 'sipping@ietf.org'
> Subject: RE: [Sipping] I-D 
> ACTION:draft-elwell-sipping-identity-interworki ng-00.txt
> 
> 
> This document has been produced in ECMA, and comments of the 
> SIP community are invited prior to finalisation of the ECMA 
> publication during the coming weeks.
> 
> john.elwell@siemens.com
> 
> 
> > -----Original Message-----
> > From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> > Sent: 29 May 2003 12:20
> > Cc: sipping@ietf.org
> > Subject: [Sipping] I-D 
> > ACTION:draft-elwell-sipping-identity-interworking-00.txt
> > 
> > 
> > A New Internet-Draft is available from the on-line
> > Internet-Drafts directories.
> > 
> > 
> > 	Title		: User identification in a SIP/QSIG environment
> > 	Author(s)	: J. Elwell
> > 	Filename	: 
> > draft-elwell-sipping-identity-interworking-00.txt
> > 	Pages		: 31
> > 	Date		: 2003-5-28
> > 	
> > This document examines means of identifying or naming users of
> > telephony services within an enterprise. Numeric names 
> (numbers) are 
> > used in traditional Private Integrated Services Networks (PISNs) 
> > using QSIG as the network signalling protocol. They are 
> also used for 
> > external communication, e.g., with a public Integrated Services 
> > Digital Network (ISDN). Names need not be numeric in Internet 
> > Protocol (IP) networks employing signalling protocols such as the 
> > Session Initiation Protocol (SIP). This document therefore looks at 
> > naming schemes that are appropriate within enterprise IP 
> networks, in 
> > particular enterprise IP networks employing SIP as the signalling 
> > protocol. It also investigates naming schemes that are 
> appropriate in 
> > a mixed QSIG/SIP enterprise network and the treatment of 
> names at an 
> > interworking point. It details the use of names not only for 
> > selecting a user to participate in a call, but also as a means of 
> > identifying a user in a call to other users in that call. ENUM and 
> > private ENUM-like services are also examined.
> > 
> > A URL for this Internet-Draft is: 
> > http://www.ietf.org/internet-drafts/draft-elwell-sipping-ident
> ity-interworking-00.txt
> 
> To remove yourself from the IETF Announcement list, send a message to 
> ietf-announce-request with the word unsubscribe in the body 
> of the message.
> 
> Internet-Drafts are also available by anonymous FTP. Login 
> with the username "anonymous" and a password of your e-mail 
> address. After logging in, type "cd internet-drafts" and then
> 	"get draft-elwell-sipping-identity-interworking-00.txt".
> 
> A list of Internet-Drafts directories can be found in 
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Internet-Draft. _______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
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Use sip-implementors@cs.columbia.edu for questions on current sip Use
sip@ietf.org for new developments of core SIP

------_=_NextPart_001_01C32617.BE1C523A
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Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2656.31">
<TITLE>RE: [Sipping] I-D =
ACTION:draft-elwell-sipping-identity-interworki ng-00.txt</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Hi John,</FONT>
</P>

<P><FONT SIZE=3D2>This is a very useful document. </FONT>
</P>

<P><FONT SIZE=3D2>In any case, I've got a few comments:</FONT>
</P>

<P><FONT SIZE=3D2>- I would like to see a mention of how to relate SIP =
URIs and tel URIs</FONT>
<BR><FONT SIZE=3D2>&nbsp; for QSIG phone numbers by putting an explicit =
reference to section</FONT>
<BR><FONT SIZE=3D2>&nbsp; 19.1.6 of RFC 3261. This could be somewhere =
in your draft, perhaps in</FONT>
<BR><FONT SIZE=3D2>&nbsp; section 11.2.2.</FONT>
</P>

<P><FONT SIZE=3D2>- There doesn't seem to be much guidance in your =
draft of how to </FONT>
<BR><FONT SIZE=3D2>&nbsp; indicate QSIG type of number information =
using the phone-context.</FONT>
<BR><FONT SIZE=3D2>&nbsp; This is precisely the type of thing that the =
phone-context was </FONT>
<BR><FONT SIZE=3D2>&nbsp; created to do for &quot;local&quot; numbers =
(i.e., within a private domain).</FONT>
<BR><FONT SIZE=3D2>&nbsp; For example, a enterprise foo.com might =
decide to use the following </FONT>
<BR><FONT SIZE=3D2>&nbsp; mapping for NPI/TON to phone-context:</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; NPI=3DPrivate =
ISO/IEC 11571, TON=3D phone-context=3D</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
-----------------------------------------------------</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>level 2 =
regional number &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; level2.foo.com</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>level 1 =
regional number &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; level1.foo.com</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>local =
(level 0) number&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; level0.foo.com</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>PISN-specific number&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; special.foo.com</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>unknown =
number&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; unknown.foo.com</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp; Of course, an enterprise could choose to use a =
phone-context of foo.com</FONT>
<BR><FONT SIZE=3D2>&nbsp; as the highest level of the phone number =
hierarchy, be it Level 2 or Level 1.</FONT>
<BR><FONT SIZE=3D2>&nbsp; Also, the host could be diffent that the =
domain in the phone-context: you</FONT>
<BR><FONT SIZE=3D2>&nbsp; could have foo.com in the phone-context and =
serviceprovider.net as the host</FONT>
<BR><FONT SIZE=3D2>&nbsp; domain. Also, this is just an example: the =
idea is that the enterprise</FONT>
<BR><FONT SIZE=3D2>&nbsp; would assign diffent phone contextes to =
different QSIG types of number.</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp; This would yield SIP URIs like </FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>sip:53756;phone-context=3Dlevel0.foo.com@foo.com;user=3Dphone</=
FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>sip:265-3756;phone-context=3Dlevel1.foo.com@foo.com;user=3Dphon=
e</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>sip:6-911;phone-context=3Dsecial.foo.com@foo.com;user=3Dphone</=
FONT>
<BR><FONT SIZE=3D2>&nbsp; </FONT>
<BR><FONT SIZE=3D2>&nbsp; This is probably something that should also =
go in section 11.2.2.</FONT>
<BR><FONT SIZE=3D2>&nbsp; </FONT>
<BR><FONT SIZE=3D2>&nbsp; All this is very much in line with =
draft-antti-rfc2806bis-08.</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Elwell, John [<A =
HREF=3D"mailto:john.elwell@siemens.com">mailto:john.elwell@siemens.com</=
A>] </FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Thursday, May 29, 2003 07:38</FONT>
<BR><FONT SIZE=3D2>&gt; To: 'sipping@ietf.org'</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: [Sipping] I-D </FONT>
<BR><FONT SIZE=3D2>&gt; ACTION:draft-elwell-sipping-identity-interworki =
ng-00.txt</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; This document has been produced in ECMA, and =
comments of the </FONT>
<BR><FONT SIZE=3D2>&gt; SIP community are invited prior to finalisation =
of the ECMA </FONT>
<BR><FONT SIZE=3D2>&gt; publication during the coming weeks.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; john.elwell@siemens.com</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; From: Internet-Drafts@ietf.org [<A =
HREF=3D"mailto:Internet-Drafts@ietf.org">mailto:Internet-Drafts@ietf.org=
</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Sent: 29 May 2003 12:20</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Cc: sipping@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Subject: [Sipping] I-D </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; =
ACTION:draft-elwell-sipping-identity-interworking-00.txt</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; A New Internet-Draft is available from the =
on-line</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Internet-Drafts directories.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &nbsp;&nbsp;&nbsp; Title&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : User identification in a =
SIP/QSIG environment</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &nbsp;&nbsp;&nbsp; =
Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : J. Elwell</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &nbsp;&nbsp;&nbsp; =
Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; =
draft-elwell-sipping-identity-interworking-00.txt</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &nbsp;&nbsp;&nbsp; Pages&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 31</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &nbsp;&nbsp;&nbsp; Date&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 2003-5-28</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; This document examines means of =
identifying or naming users of</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; telephony services within an enterprise. =
Numeric names </FONT>
<BR><FONT SIZE=3D2>&gt; (numbers) are </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; used in traditional Private Integrated =
Services Networks (PISNs) </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; using QSIG as the network signalling =
protocol. They are </FONT>
<BR><FONT SIZE=3D2>&gt; also used for </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; external communication, e.g., with a =
public Integrated Services </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Digital Network (ISDN). Names need not be =
numeric in Internet </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Protocol (IP) networks employing =
signalling protocols such as the </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Session Initiation Protocol (SIP). This =
document therefore looks at </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; naming schemes that are appropriate within =
enterprise IP </FONT>
<BR><FONT SIZE=3D2>&gt; networks, in </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; particular enterprise IP networks =
employing SIP as the signalling </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; protocol. It also investigates naming =
schemes that are </FONT>
<BR><FONT SIZE=3D2>&gt; appropriate in </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; a mixed QSIG/SIP enterprise network and =
the treatment of </FONT>
<BR><FONT SIZE=3D2>&gt; names at an </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; interworking point. It details the use of =
names not only for </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; selecting a user to participate in a call, =
but also as a means of </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; identifying a user in a call to other =
users in that call. ENUM and </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; private ENUM-like services are also =
examined.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; A URL for this Internet-Draft is: </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; <A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-elwell-sipping-ident" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-elwell-sippi=
ng-ident</A></FONT>
<BR><FONT SIZE=3D2>&gt; ity-interworking-00.txt</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; To remove yourself from the IETF Announcement =
list, send a message to </FONT>
<BR><FONT SIZE=3D2>&gt; ietf-announce-request with the word unsubscribe =
in the body </FONT>
<BR><FONT SIZE=3D2>&gt; of the message.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Internet-Drafts are also available by anonymous =
FTP. Login </FONT>
<BR><FONT SIZE=3D2>&gt; with the username &quot;anonymous&quot; and a =
password of your e-mail </FONT>
<BR><FONT SIZE=3D2>&gt; address. After logging in, type &quot;cd =
internet-drafts&quot; and then</FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;get =
draft-elwell-sipping-identity-interworking-00.txt&quot;.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; A list of Internet-Drafts directories can be =
found in </FONT>
<BR><FONT SIZE=3D2><A HREF=3D"http://www.ietf.org/shadow.html" =
TARGET=3D"_blank">http://www.ietf.org/shadow.html</A> </FONT>
<BR><FONT SIZE=3D2>or <A =
HREF=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt" =
TARGET=3D"_blank">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</A></FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Internet-Drafts can also be obtained by =
e-mail.</FONT>
</P>

<P><FONT SIZE=3D2>Send a message to:</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>mailserv@ietf.org.</FONT>
<BR><FONT SIZE=3D2>In the body type:</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>&quot;FILE =
/internet-drafts/draft-elwell-sipping-identity-interworking-00.txt&quot;=
.</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR><FONT SIZE=3D2>NOTE:&nbsp;&nbsp; The mail server at ietf.org can =
return the document in</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>MIME-encoded form by using the &quot;mpack&quot; =
utility.&nbsp; To use this</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>feature, =
insert the command &quot;ENCODING mime&quot; before the =
&quot;FILE&quot;</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>command.&nbsp; To decode the response(s), you will need =
&quot;munpack&quot; or</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>a =
MIME-compliant mail reader.&nbsp; Different MIME-compliant mail =
readers</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>exhibit =
different behavior, especially when dealing with</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>&quot;multipart&quot; MIME messages (i.e. documents which have =
been split</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>up into =
multiple messages), so check your local documentation on</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>how to =
manipulate these messages.</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR><FONT SIZE=3D2>Below is the data which will enable a MIME compliant =
mail reader implementation to automatically retrieve the ASCII version =
of the Internet-Draft. =
_______________________________________________</FONT></P>

<P><FONT SIZE=3D2>Sipping mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sipping" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sipping</A></FO=
NT>
<BR><FONT SIZE=3D2>This list is for NEW development of the application =
of SIP</FONT>
<BR><FONT SIZE=3D2>Use sip-implementors@cs.columbia.edu for questions =
on current sip Use sip@ietf.org for new developments of core SIP</FONT>
</P>

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From mailnull@www1.ietf.org  Thu May 29 18:34:02 2003
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Message-ID: <A3851AA1B761E944912B20D1E95A7EFE08B38E@radvpost.radvision.com>
From: Orit Levin <orit@radvision.com>
To: "'Eric Burger'" <eburger@snowshore.com>
Cc: "Sipping (E-mail)" <sipping@ietf.org>
Date: Thu, 29 May 2003 18:19:54 -0400
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Subject: [Sipping] conf=comments on draft-burger-sipping-netann-05
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Hi Eric!
I have a number of questions/comments on the draft:

1. I understand that the draft purpose is to become an informational RFC.
What stage of the process are we at?
2. The draft uses MUST, SHOULD, MAY, etc. Is it the right language for an
informational RFC? (Also, please, take a look at the section 5.1 which says
"This section is informative" :-)
3. Section 2. "Mechanism" discusses the need for instantiation of a service
and the corresponding syntax as "the concatenation of the service name and
an optional service instance identifier, separated by an equal sign". It
also brings up "multiple, separate conference sessions" as an example for a
service instantiation.
I endorse this direction and find it completely compatible and fitting the
more general approach of the draft-sipping-cc-conferencing.
4. That said, in section 5 "Conference Service" a different syntax is
defined!!! It introduces a single "conf-id" parameter to be inserted as a
global SIP URI parameter - not in the USER portion of the URI... I find this
approach dangerous (even as informational) because it doesn't add any
functionality (to specifying a "service instance" in the USER part of the
URI) and will lead to interoperability problems (because it is different
from the approach taken by both the conferencing-framework and the
cc-conferencing WG documents).

Any thoughts?

Thanks,
Orit Levin
Chief Architect
RADVISION
Phone: +1.201.6896330

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A New Internet-Draft is available from the on-line Internet-Drafts direct=
ories.


	Title		: Requirements for Full-mesh Conference Model using=20
                          Baseline Session Initiation Protocol
	Author(s)	: V. Paulsamy
	Filename	: draft-victor-fullmesh-reqts-00.txt
	Pages		: 7
	Date		: 2003-5-29
=09
This memo enumerates requirements to realize full-mesh multiparty=20
conferencing using baseline Session Initiation Protocol. Primary=20
requirement is to advertise conference members=C6 addresses to new=20
participants by concatenating multiple SDP session descriptions=20
together. This particular requirement can solve several well-known=20
problems, including disjoint meshes and race conditions, that are=20
lingering in full-mesh conferencing for a while now.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-victor-fullmesh-reqts-00.txt

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Date: Fri, 30 May 2003 14:52:17 +0100
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sipping] I-D ACTION:draft-elwell-sipping-identity-interworki
	ng-00.txt
To: "'Francois Audet'" <audet@nortelnetworks.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>
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Francois,
 
Thanks for your comments. Concerning your point about phone-context, I am
not sure that this parameter is defined for a SIP URI (as opposed to a Tel
URL). The only reference I can find in RFC3261 is an example at the end of
19.1.6. I agree that your proposed use of this parameter might be useful,
but I am reluctant to propose anything without a clearer indication that
this is indeed a standard part of the SIP URI. Any other opinions?
 
John
 
John Elwell - Technical Consultant 
Tel: +44 (0)115 943 4989 
Fax: +44 (0)115 943 4969 
e-mail: mailto:john.elwell@siemens.com <mailto:john.elwell@siemens.com>  
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<http://www.siemens-design-services.com/>  

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-----Original Message-----
From: Francois Audet [mailto:audet@nortelnetworks.com]
Sent: 29 May 2003 20:23
To: 'Elwell, John'; 'sipping@ietf.org'
Subject: RE: [Sipping] I-D ACTION:draft-elwell-sipping-identity-interworki
ng-00.txt



Hi John, 

This is a very useful document. 

In any case, I've got a few comments: 

- I would like to see a mention of how to relate SIP URIs and tel URIs 
  for QSIG phone numbers by putting an explicit reference to section 
  19.1.6 of RFC 3261. This could be somewhere in your draft, perhaps in 
  section 11.2.2. 

- There doesn't seem to be much guidance in your draft of how to 
  indicate QSIG type of number information using the phone-context. 
  This is precisely the type of thing that the phone-context was 
  created to do for "local" numbers (i.e., within a private domain). 
  For example, a enterprise foo.com might decide to use the following 
  mapping for NPI/TON to phone-context: 

        NPI=Private ISO/IEC 11571, TON= phone-context= 
      ----------------------------------------------------- 
        level 2 regional number                 level2.foo.com 
        level 1 regional number                 level1.foo.com 
        local (level 0) number                  level0.foo.com 
        PISN-specific number                    special.foo.com 
        unknown number                          unknown.foo.com 

  Of course, an enterprise could choose to use a phone-context of foo.com 
  as the highest level of the phone number hierarchy, be it Level 2 or Level
1. 
  Also, the host could be diffent that the domain in the phone-context: you 
  could have foo.com in the phone-context and serviceprovider.net as the
host 
  domain. Also, this is just an example: the idea is that the enterprise 
  would assign diffent phone contextes to different QSIG types of number. 

  This would yield SIP URIs like 
        sip:53756;phone-context=level0.foo.com@foo.com;user=phone 
        sip:265-3756;phone-context=level1.foo.com@foo.com;user=phone 
        sip:6-911;phone-context=secial.foo.com@foo.com;user=phone 
  
  This is probably something that should also go in section 11.2.2. 
  
  All this is very much in line with draft-antti-rfc2806bis-08. 

> -----Original Message----- 
> From: Elwell, John [ mailto:john.elwell@siemens.com
<mailto:john.elwell@siemens.com> ] 
> Sent: Thursday, May 29, 2003 07:38 
> To: 'sipping@ietf.org' 
> Subject: RE: [Sipping] I-D 
> ACTION:draft-elwell-sipping-identity-interworki ng-00.txt 
> 
> 
> This document has been produced in ECMA, and comments of the 
> SIP community are invited prior to finalisation of the ECMA 
> publication during the coming weeks. 
> 
> john.elwell@siemens.com 
> 
> 
> > -----Original Message----- 
> > From: Internet-Drafts@ietf.org [ mailto:Internet-Drafts@ietf.org
<mailto:Internet-Drafts@ietf.org> ] 
> > Sent: 29 May 2003 12:20 
> > Cc: sipping@ietf.org 
> > Subject: [Sipping] I-D 
> > ACTION:draft-elwell-sipping-identity-interworking-00.txt 
> > 
> > 
> > A New Internet-Draft is available from the on-line 
> > Internet-Drafts directories. 
> > 
> > 
> >     Title           : User identification in a SIP/QSIG environment 
> >     Author(s)       : J. Elwell 
> >     Filename        : 
> > draft-elwell-sipping-identity-interworking-00.txt 
> >     Pages           : 31 
> >     Date            : 2003-5-28 
> >     
> > This document examines means of identifying or naming users of 
> > telephony services within an enterprise. Numeric names 
> (numbers) are 
> > used in traditional Private Integrated Services Networks (PISNs) 
> > using QSIG as the network signalling protocol. They are 
> also used for 
> > external communication, e.g., with a public Integrated Services 
> > Digital Network (ISDN). Names need not be numeric in Internet 
> > Protocol (IP) networks employing signalling protocols such as the 
> > Session Initiation Protocol (SIP). This document therefore looks at 
> > naming schemes that are appropriate within enterprise IP 
> networks, in 
> > particular enterprise IP networks employing SIP as the signalling 
> > protocol. It also investigates naming schemes that are 
> appropriate in 
> > a mixed QSIG/SIP enterprise network and the treatment of 
> names at an 
> > interworking point. It details the use of names not only for 
> > selecting a user to participate in a call, but also as a means of 
> > identifying a user in a call to other users in that call. ENUM and 
> > private ENUM-like services are also examined. 
> > 
> > A URL for this Internet-Draft is: 
> > http://www.ietf.org/internet-drafts/draft-elwell-sipping-ident
<http://www.ietf.org/internet-drafts/draft-elwell-sipping-ident>  
> ity-interworking-00.txt 
> 
> To remove yourself from the IETF Announcement list, send a message to 
> ietf-announce-request with the word unsubscribe in the body 
> of the message. 
> 
> Internet-Drafts are also available by anonymous FTP. Login 
> with the username "anonymous" and a password of your e-mail 
> address. After logging in, type "cd internet-drafts" and then 
>       "get draft-elwell-sipping-identity-interworking-00.txt". 
> 
> A list of Internet-Drafts directories can be found in 
http://www.ietf.org/shadow.html <http://www.ietf.org/shadow.html>  
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
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Internet-Draft. _______________________________________________

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From mailnull@www1.ietf.org  Fri May 30 10:18:02 2003
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Subject: [Sipping] Sip/Simple
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Dear Sirs,

I would like to know if there is any sip client that is used for instant 
messaging , my search is for a Sip/Simple IM client , I have been told by my 
friend that the msn messenger 4.6 does use Sip .
Does anyone know how to connect to a sip server using msn messenger 4.6.
I have been told also that the windows messenger ( for xp ) uses sip too , 
but I have not tried it yet .
Does anyone know if there is a Simple library in C language , the Sip 
library is just beyond my need and is very big to understand and use.

Any help would be dramatically appreciated !!!

Regards.

_________________________________________________________________
The new MSN 8: smart spam protection and 2 months FREE*  
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_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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This is a cryptographically signed message in MIME format.

--------------ms020104090105060907060505
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Dear all,
	I saw the Torval proposal for authenticate SIP messages with EAP.

The IETF document draft-torvinen-http-eap-02.txt

Was that proposal been inserted in a new draft or RFC?

We are working to find an extensible authentication schema for SIP and 
are we now evaluating that old proposal.

Thank you in advance,

Corrado Derenale.

--------------ms020104090105060907060505
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From: Jari Arkko <jari.arkko@piuha.net>
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To: derenale@polito.it
Cc: vesa.torvinen@ericsson.fi, aki.niemi@nokia.com, sipping@ietf.org
Subject: Re: [Sipping] SIP and EAP
References: <3ED75A29.8070500@polito.it>
In-Reply-To: <3ED75A29.8070500@polito.it>
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Corrado Derenale wrote:
> Dear all,
>     I saw the Torval proposal for authenticate SIP messages with EAP.
> 
> The IETF document draft-torvinen-http-eap-02.txt
> 
> Was that proposal been inserted in a new draft or RFC?
> 
> We are working to find an extensible authentication schema for SIP and 
> are we now evaluating that old proposal.

This old proposal was later replaced with the use of the specifically
needed authentication scheme (UMTS AKA) in the context of Digest
framework. The reasons for this switch included the desire to
keep EAP only for network access authorization purposes, and the
desire to minimize impacts to SIP.

See also RFC 3310.

--Jari


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From: "Francois Audet" <audet@nortelnetworks.com>
To: "'Elwell, John'" <john.elwell@siemens.com>,
        "'sipping@ietf.org'"
	 <sipping@ietf.org>,
        "'Henning Schulzrinne'" <hgs@cs.columbia.edu>
Subject: RE: [Sipping] I-D ACTION:draft-elwell-sipping-identity-interworki
	 ng-00.txt
Date: Fri, 30 May 2003 10:55:51 -0700
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Hi John,

In the SIP BNF description in RFC3261, on page 223, it is stated 
that if the userinfo in the SIP-URI is a telephone-subscriber 
(as opposed to a user), then the telephone-subscriber BNF is 
defined in RFC2806. (Although not explicitely stated, I assume 
it also means that the "user=phone" user-param needs to be 
included). Perhaps Henning can confirm? 

The guidance I have described below is indeed when a
telephone-subscriber instead of a user (for the userinfo) is
used. I think it would be very useful to spell out that
type of guidance in the QSIG identity interworking draft.

Of course, it would also be doable to use the user instead of the
telephone-subscriber field in the userinfo. In that case, 
essentially any type of user@hostport format is allowed, and there
is not much guidance we can give. It then becomes a matter of 
converting phone numbers to SIP URIs. QSIG-Private/Level1=265-3756 could
be mapped to sip:2653756@foo.com, (or sip:audet@foo.com for that matter). 

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com] 
> Sent: Friday, May 30, 2003 06:52
> To: Audet, Francois [SC100:4K02:EXCH]; 'sipping@ietf.org'
> Subject: RE: [Sipping] I-D 
> ACTION:draft-elwell-sipping-identity-interworki ng-00.txt
> 
> 
> Francois,
>  
> Thanks for your comments. Concerning your point about 
> phone-context, I am not sure that this parameter is defined 
> for a SIP URI (as opposed to a Tel URL). The only reference I 
> can find in RFC3261 is an example at the end of 19.1.6. I 
> agree that your proposed use of this parameter might be 
> useful, but I am reluctant to propose anything without a 
> clearer indication that this is indeed a standard part of the 
> SIP URI. Any other opinions?
>  
> John
>  
> John Elwell - Technical Consultant 
> Tel: +44 (0)115 943 4989 
> Fax: +44 (0)115 943 4969 
> e-mail: mailto:john.elwell@siemens.com 
> <mailto:john.elwell@siemens.com>  
> web: http://www.siemens-design-services.com
> <http://www.siemens-design-services.com/>  
> 
> Siemens Communications 
> Technology Drive, Beeston, Nottingham, NG9 1LA.  England. 
> 
> "Siemens Communications - a division of Siemens plc,  
> Registered No: 727817, England. Registered office: Siemens 
> House, Oldbury, Bracknell, Berkshire, RG12 8FZ.
> 
> This communication contains information which is confidential 
> and may also be privileged. It is for the exclusive use of 
> the addressee. If you are not the addressee please note that 
> any distribution, reproduction, copying, publication or use 
> of this communication or the information is prohibited. If 
> you have received this communication in error, please contact 
> us immediately and also delete the communication from your 
> computer. We accept no liability for any loss or damage 
> suffered by any person arising from use of this e-mail."
> 
>  
> 
> -----Original Message-----
> From: Francois Audet [mailto:audet@nortelnetworks.com]
> Sent: 29 May 2003 20:23
> To: 'Elwell, John'; 'sipping@ietf.org'
> Subject: RE: [Sipping] I-D 
> ACTION:draft-elwell-sipping-identity-interworki
> ng-00.txt
> 
> 
> 
> Hi John, 
> 
> This is a very useful document. 
> 
> In any case, I've got a few comments: 
> 
> - I would like to see a mention of how to relate SIP URIs and 
> tel URIs 
>   for QSIG phone numbers by putting an explicit reference to section 
>   19.1.6 of RFC 3261. This could be somewhere in your draft, 
> perhaps in 
>   section 11.2.2. 
> 
> - There doesn't seem to be much guidance in your draft of how to 
>   indicate QSIG type of number information using the phone-context. 
>   This is precisely the type of thing that the phone-context was 
>   created to do for "local" numbers (i.e., within a private domain). 
>   For example, a enterprise foo.com might decide to use the following 
>   mapping for NPI/TON to phone-context: 
> 
>         NPI=Private ISO/IEC 11571, TON= phone-context= 
>       ----------------------------------------------------- 
>         level 2 regional number                 level2.foo.com 
>         level 1 regional number                 level1.foo.com 
>         local (level 0) number                  level0.foo.com 
>         PISN-specific number                    special.foo.com 
>         unknown number                          unknown.foo.com 
> 
>   Of course, an enterprise could choose to use a 
> phone-context of foo.com 
>   as the highest level of the phone number hierarchy, be it 
> Level 2 or Level 1. 
>   Also, the host could be diffent that the domain in the 
> phone-context: you 
>   could have foo.com in the phone-context and 
> serviceprovider.net as the host 
>   domain. Also, this is just an example: the idea is that the 
> enterprise 
>   would assign diffent phone contextes to different QSIG 
> types of number. 
> 
>   This would yield SIP URIs like 
>         sip:53756;phone-context=level0.foo.com@foo.com;user=phone 
>         sip:265-3756;phone-context=level1.foo.com@foo.com;user=phone 
>         sip:6-911;phone-context=secial.foo.com@foo.com;user=phone 
>   
>   This is probably something that should also go in section 11.2.2. 
>   
>   All this is very much in line with draft-antti-rfc2806bis-08. 
> 
> > -----Original Message-----
> > From: Elwell, John [ mailto:john.elwell@siemens.com
> <mailto:john.elwell@siemens.com> ] 
> > Sent: Thursday, May 29, 2003 07:38
> > To: 'sipping@ietf.org' 
> > Subject: RE: [Sipping] I-D 
> > ACTION:draft-elwell-sipping-identity-interworki ng-00.txt 
> > 
> > 
> > This document has been produced in ECMA, and comments of the
> > SIP community are invited prior to finalisation of the ECMA 
> > publication during the coming weeks. 
> > 
> > john.elwell@siemens.com
> > 
> > 
> > > -----Original Message-----
> > > From: Internet-Drafts@ietf.org [ mailto:Internet-Drafts@ietf.org
> <mailto:Internet-Drafts@ietf.org> ] 
> > > Sent: 29 May 2003 12:20
> > > Cc: sipping@ietf.org 
> > > Subject: [Sipping] I-D 
> > > ACTION:draft-elwell-sipping-identity-interworking-00.txt 
> > > 
> > > 
> > > A New Internet-Draft is available from the on-line
> > > Internet-Drafts directories. 
> > > 
> > > 
> > >     Title           : User identification in a SIP/QSIG 
> environment 
> > >     Author(s)       : J. Elwell 
> > >     Filename        : 
> > > draft-elwell-sipping-identity-interworking-00.txt 
> > >     Pages           : 31 
> > >     Date            : 2003-5-28 
> > >     
> > > This document examines means of identifying or naming users of
> > > telephony services within an enterprise. Numeric names 
> > (numbers) are
> > > used in traditional Private Integrated Services Networks (PISNs)
> > > using QSIG as the network signalling protocol. They are 
> > also used for
> > > external communication, e.g., with a public Integrated Services
> > > Digital Network (ISDN). Names need not be numeric in Internet 
> > > Protocol (IP) networks employing signalling protocols such as the 
> > > Session Initiation Protocol (SIP). This document 
> therefore looks at 
> > > naming schemes that are appropriate within enterprise IP 
> > networks, in
> > > particular enterprise IP networks employing SIP as the signalling
> > > protocol. It also investigates naming schemes that are 
> > appropriate in
> > > a mixed QSIG/SIP enterprise network and the treatment of
> > names at an
> > > interworking point. It details the use of names not only for
> > > selecting a user to participate in a call, but also as a means of 
> > > identifying a user in a call to other users in that call. 
> ENUM and 
> > > private ENUM-like services are also examined. 
> > > 
> > > A URL for this Internet-Draft is:
> > > http://www.ietf.org/internet-drafts/draft-elwell-sipping-ident
> <http://www.ietf.org/internet-drafts/draft-elwell-sipping-ident>  
> > ity-interworking-00.txt
> > 
> > To remove yourself from the IETF Announcement list, send a 
> message to
> > ietf-announce-request with the word unsubscribe in the body 
> > of the message. 
> > 
> > Internet-Drafts are also available by anonymous FTP. Login
> > with the username "anonymous" and a password of your e-mail 
> > address. After logging in, type "cd internet-drafts" and then 
> >       "get draft-elwell-sipping-identity-interworking-00.txt". 
> > 
> > A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html <http://www.ietf.org/shadow.html>  
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> <ftp://ftp.ietf.org/ietf/1shadow-sites.txt>  
> 
> 
> Internet-Drafts can also be obtained by e-mail. 
> 
> Send a message to: 
>         mailserv@ietf.org. 
> In the body type: 
>         "FILE 
> /internet-drafts/draft-elwell-sipping-identity-interworking-00.txt". 
>         
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> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
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> This list is for NEW development of the application of SIP 
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sip@ietf.org for new developments of core SIP 
> 
> 

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<TITLE>RE: [Sipping] I-D =
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<P><FONT SIZE=3D2>Hi John,</FONT>
</P>

<P><FONT SIZE=3D2>In the SIP BNF description in RFC3261, on page 223, =
it is stated </FONT>
<BR><FONT SIZE=3D2>that if the userinfo in the SIP-URI is a =
telephone-subscriber </FONT>
<BR><FONT SIZE=3D2>(as opposed to a user), then the =
telephone-subscriber BNF is </FONT>
<BR><FONT SIZE=3D2>defined in RFC2806. (Although not explicitely =
stated, I assume </FONT>
<BR><FONT SIZE=3D2>it also means that the &quot;user=3Dphone&quot; =
user-param needs to be </FONT>
<BR><FONT SIZE=3D2>included). Perhaps Henning can confirm? </FONT>
</P>

<P><FONT SIZE=3D2>The guidance I have described below is indeed when =
a</FONT>
<BR><FONT SIZE=3D2>telephone-subscriber instead of a user (for the =
userinfo) is</FONT>
<BR><FONT SIZE=3D2>used. I think it would be very useful to spell out =
that</FONT>
<BR><FONT SIZE=3D2>type of guidance in the QSIG identity interworking =
draft.</FONT>
</P>

<P><FONT SIZE=3D2>Of course, it would also be doable to use the user =
instead of the</FONT>
<BR><FONT SIZE=3D2>telephone-subscriber field in the userinfo. In that =
case, </FONT>
<BR><FONT SIZE=3D2>essentially any type of user@hostport format is =
allowed, and there</FONT>
<BR><FONT SIZE=3D2>is not much guidance we can give. It then becomes a =
matter of </FONT>
<BR><FONT SIZE=3D2>converting phone numbers to SIP URIs. =
QSIG-Private/Level1=3D265-3756 could</FONT>
<BR><FONT SIZE=3D2>be mapped to sip:2653756@foo.com, (or =
sip:audet@foo.com for that matter). </FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Elwell, John [<A =
HREF=3D"mailto:john.elwell@siemens.com">mailto:john.elwell@siemens.com</=
A>] </FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Friday, May 30, 2003 06:52</FONT>
<BR><FONT SIZE=3D2>&gt; To: Audet, Francois [SC100:4K02:EXCH]; =
'sipping@ietf.org'</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: [Sipping] I-D </FONT>
<BR><FONT SIZE=3D2>&gt; ACTION:draft-elwell-sipping-identity-interworki =
ng-00.txt</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Francois,</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; Thanks for your comments. Concerning your point =
about </FONT>
<BR><FONT SIZE=3D2>&gt; phone-context, I am not sure that this =
parameter is defined </FONT>
<BR><FONT SIZE=3D2>&gt; for a SIP URI (as opposed to a Tel URL). The =
only reference I </FONT>
<BR><FONT SIZE=3D2>&gt; can find in RFC3261 is an example at the end of =
19.1.6. I </FONT>
<BR><FONT SIZE=3D2>&gt; agree that your proposed use of this parameter =
might be </FONT>
<BR><FONT SIZE=3D2>&gt; useful, but I am reluctant to propose anything =
without a </FONT>
<BR><FONT SIZE=3D2>&gt; clearer indication that this is indeed a =
standard part of the </FONT>
<BR><FONT SIZE=3D2>&gt; SIP URI. Any other opinions?</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; John</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; John Elwell - Technical Consultant </FONT>
<BR><FONT SIZE=3D2>&gt; Tel: +44 (0)115 943 4989 </FONT>
<BR><FONT SIZE=3D2>&gt; Fax: +44 (0)115 943 4969 </FONT>
<BR><FONT SIZE=3D2>&gt; e-mail: <A =
HREF=3D"mailto:john.elwell@siemens.com">mailto:john.elwell@siemens.com</=
A> </FONT>
<BR><FONT SIZE=3D2>&gt; &lt;<A =
HREF=3D"mailto:john.elwell@siemens.com">mailto:john.elwell@siemens.com</=
A>&gt;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; web: <A =
HREF=3D"http://www.siemens-design-services.com" =
TARGET=3D"_blank">http://www.siemens-design-services.com</A></FONT>
<BR><FONT SIZE=3D2>&gt; &lt;<A =
HREF=3D"http://www.siemens-design-services.com/" =
TARGET=3D"_blank">http://www.siemens-design-services.com/</A>&gt;&nbsp; =
</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Siemens Communications </FONT>
<BR><FONT SIZE=3D2>&gt; Technology Drive, Beeston, Nottingham, NG9 =
1LA.&nbsp; England. </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &quot;Siemens Communications - a division of =
Siemens plc,&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; Registered No: 727817, England. Registered =
office: Siemens </FONT>
<BR><FONT SIZE=3D2>&gt; House, Oldbury, Bracknell, Berkshire, RG12 =
8FZ.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; This communication contains information which =
is confidential </FONT>
<BR><FONT SIZE=3D2>&gt; and may also be privileged. It is for the =
exclusive use of </FONT>
<BR><FONT SIZE=3D2>&gt; the addressee. If you are not the addressee =
please note that </FONT>
<BR><FONT SIZE=3D2>&gt; any distribution, reproduction, copying, =
publication or use </FONT>
<BR><FONT SIZE=3D2>&gt; of this communication or the information is =
prohibited. If </FONT>
<BR><FONT SIZE=3D2>&gt; you have received this communication in error, =
please contact </FONT>
<BR><FONT SIZE=3D2>&gt; us immediately and also delete the =
communication from your </FONT>
<BR><FONT SIZE=3D2>&gt; computer. We accept no liability for any loss =
or damage </FONT>
<BR><FONT SIZE=3D2>&gt; suffered by any person arising from use of this =
e-mail.&quot;</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Francois Audet [<A =
HREF=3D"mailto:audet@nortelnetworks.com">mailto:audet@nortelnetworks.com=
</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: 29 May 2003 20:23</FONT>
<BR><FONT SIZE=3D2>&gt; To: 'Elwell, John'; 'sipping@ietf.org'</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: [Sipping] I-D </FONT>
<BR><FONT SIZE=3D2>&gt; =
ACTION:draft-elwell-sipping-identity-interworki</FONT>
<BR><FONT SIZE=3D2>&gt; ng-00.txt</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Hi John, </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; This is a very useful document. </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; In any case, I've got a few comments: </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; - I would like to see a mention of how to =
relate SIP URIs and </FONT>
<BR><FONT SIZE=3D2>&gt; tel URIs </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; for QSIG phone numbers by putting =
an explicit reference to section </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; 19.1.6 of RFC 3261. This could be =
somewhere in your draft, </FONT>
<BR><FONT SIZE=3D2>&gt; perhaps in </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; section 11.2.2. </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; - There doesn't seem to be much guidance in =
your draft of how to </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; indicate QSIG type of number =
information using the phone-context. </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; This is precisely the type of thing =
that the phone-context was </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; created to do for &quot;local&quot; =
numbers (i.e., within a private domain). </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; For example, a enterprise foo.com =
might decide to use the following </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; mapping for NPI/TON to =
phone-context: </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
NPI=3DPrivate ISO/IEC 11571, TON=3D phone-context=3D </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
----------------------------------------------------- </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
level 2 regional =
number&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; level2.foo.com </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
level 1 regional =
number&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; level1.foo.com </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
local (level 0) =
number&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; level0.foo.com </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
PISN-specific =
number&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; special.foo.com =
</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
unknown =
number&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp; unknown.foo.com </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; Of course, an enterprise could =
choose to use a </FONT>
<BR><FONT SIZE=3D2>&gt; phone-context of foo.com </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; as the highest level of the phone =
number hierarchy, be it </FONT>
<BR><FONT SIZE=3D2>&gt; Level 2 or Level 1. </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; Also, the host could be diffent =
that the domain in the </FONT>
<BR><FONT SIZE=3D2>&gt; phone-context: you </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; could have foo.com in the =
phone-context and </FONT>
<BR><FONT SIZE=3D2>&gt; serviceprovider.net as the host </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; domain. Also, this is just an =
example: the idea is that the </FONT>
<BR><FONT SIZE=3D2>&gt; enterprise </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; would assign diffent phone =
contextes to different QSIG </FONT>
<BR><FONT SIZE=3D2>&gt; types of number. </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; This would yield SIP URIs like =
</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
sip:53756;phone-context=3Dlevel0.foo.com@foo.com;user=3Dphone </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
sip:265-3756;phone-context=3Dlevel1.foo.com@foo.com;user=3Dphone =
</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
sip:6-911;phone-context=3Dsecial.foo.com@foo.com;user=3Dphone </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; This is probably something that =
should also go in section 11.2.2. </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp; All this is very much in line with =
draft-antti-rfc2806bis-08. </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; From: Elwell, John [ <A =
HREF=3D"mailto:john.elwell@siemens.com">mailto:john.elwell@siemens.com</=
A></FONT>
<BR><FONT SIZE=3D2>&gt; &lt;<A =
HREF=3D"mailto:john.elwell@siemens.com">mailto:john.elwell@siemens.com</=
A>&gt; ] </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Sent: Thursday, May 29, 2003 07:38</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; To: 'sipping@ietf.org' </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Subject: RE: [Sipping] I-D </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; =
ACTION:draft-elwell-sipping-identity-interworki ng-00.txt </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; This document has been produced in ECMA, =
and comments of the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; SIP community are invited prior to =
finalisation of the ECMA </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; publication during the coming weeks. =
</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; john.elwell@siemens.com</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; From: Internet-Drafts@ietf.org [ <A =
HREF=3D"mailto:Internet-Drafts@ietf.org">mailto:Internet-Drafts@ietf.org=
</A></FONT>
<BR><FONT SIZE=3D2>&gt; &lt;<A =
HREF=3D"mailto:Internet-Drafts@ietf.org">mailto:Internet-Drafts@ietf.org=
</A>&gt; ] </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Sent: 29 May 2003 12:20</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Cc: sipping@ietf.org </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Subject: [Sipping] I-D </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; =
ACTION:draft-elwell-sipping-identity-interworking-00.txt </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; A New Internet-Draft is available =
from the on-line</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Internet-Drafts directories. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; =
Title&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
User identification in a SIP/QSIG </FONT>
<BR><FONT SIZE=3D2>&gt; environment </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; =
Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : J. Elwell </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; =
Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; =
draft-elwell-sipping-identity-interworking-00.txt </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; =
Pages&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 31 =
</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; =
Date&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
: 2003-5-28 </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; This document examines means of =
identifying or naming users of</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; telephony services within an =
enterprise. Numeric names </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; (numbers) are</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; used in traditional Private =
Integrated Services Networks (PISNs)</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; using QSIG as the network signalling =
protocol. They are </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; also used for</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; external communication, e.g., with a =
public Integrated Services</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Digital Network (ISDN). Names need =
not be numeric in Internet </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Protocol (IP) networks employing =
signalling protocols such as the </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Session Initiation Protocol (SIP). =
This document </FONT>
<BR><FONT SIZE=3D2>&gt; therefore looks at </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; naming schemes that are appropriate =
within enterprise IP </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; networks, in</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; particular enterprise IP networks =
employing SIP as the signalling</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; protocol. It also investigates naming =
schemes that are </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; appropriate in</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; a mixed QSIG/SIP enterprise network =
and the treatment of</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; names at an</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; interworking point. It details the =
use of names not only for</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; selecting a user to participate in a =
call, but also as a means of </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; identifying a user in a call to other =
users in that call. </FONT>
<BR><FONT SIZE=3D2>&gt; ENUM and </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; private ENUM-like services are also =
examined. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; A URL for this Internet-Draft =
is:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; <A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-elwell-sipping-ident" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-elwell-sippi=
ng-ident</A></FONT>
<BR><FONT SIZE=3D2>&gt; &lt;<A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-elwell-sipping-ident" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-elwell-sippi=
ng-ident</A>&gt;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; ity-interworking-00.txt</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; To remove yourself from the IETF =
Announcement list, send a </FONT>
<BR><FONT SIZE=3D2>&gt; message to</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; ietf-announce-request with the word =
unsubscribe in the body </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; of the message. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Internet-Drafts are also available by =
anonymous FTP. Login</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; with the username &quot;anonymous&quot; =
and a password of your e-mail </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; address. After logging in, type &quot;cd =
internet-drafts&quot; and then </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&quot;get draft-elwell-sipping-identity-interworking-00.txt&quot;. =
</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; A list of Internet-Drafts directories can =
be found in</FONT>
<BR><FONT SIZE=3D2>&gt; <A HREF=3D"http://www.ietf.org/shadow.html" =
TARGET=3D"_blank">http://www.ietf.org/shadow.html</A> &lt;<A =
HREF=3D"http://www.ietf.org/shadow.html" =
TARGET=3D"_blank">http://www.ietf.org/shadow.html</A>&gt;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; or <A =
HREF=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt" =
TARGET=3D"_blank">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</A></FONT>
<BR><FONT SIZE=3D2>&gt; &lt;<A =
HREF=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt" =
TARGET=3D"_blank">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</A>&gt;&nbsp=
; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Internet-Drafts can also be obtained by e-mail. =
</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Send a message to: </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
mailserv@ietf.org. </FONT>
<BR><FONT SIZE=3D2>&gt; In the body type: </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&quot;FILE </FONT>
<BR><FONT SIZE=3D2>&gt; =
/internet-drafts/draft-elwell-sipping-identity-interworking-00.txt&quot;=
. </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</FONT>
<BR><FONT SIZE=3D2>&gt; NOTE:&nbsp;&nbsp; The mail server at ietf.org =
can return the document in </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
MIME-encoded form by using the &quot;mpack&quot; utility.&nbsp; To use =
this </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
feature, insert the command &quot;ENCODING mime&quot; before the =
&quot;FILE&quot; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
command.&nbsp; To decode the response(s), you will need </FONT>
<BR><FONT SIZE=3D2>&gt; &quot;munpack&quot; or </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
a MIME-compliant mail reader.&nbsp; Different </FONT>
<BR><FONT SIZE=3D2>&gt; MIME-compliant mail readers</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
exhibit different behavior, especially when dealing with </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&quot;multipart&quot; MIME messages (i.e. documents which have </FONT>
<BR><FONT SIZE=3D2>&gt; been split </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
up into multiple messages), so check your local </FONT>
<BR><FONT SIZE=3D2>&gt; documentation on </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
how to manipulate these messages. </FONT>
<BR><FONT =
SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT =
SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; Below is the data which will enable a MIME =
compliant mail </FONT>
<BR><FONT SIZE=3D2>&gt; reader implementation to automatically retrieve =
the ASCII </FONT>
<BR><FONT SIZE=3D2>&gt; version of the Internet-Draft. </FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Sipping mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sipping" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sipping</A></FO=
NT>
<BR><FONT SIZE=3D2>&gt; &lt;<A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sipping" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sipping</A>&gt;=
&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the =
application of SIP </FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current </FONT>
<BR><FONT SIZE=3D2>&gt; sip Use sip@ietf.org for new developments of =
core SIP </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: Jonathan Lennox <lennox@cs.columbia.edu>
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Date: Fri, 30 May 2003 14:51:55 -0400
To: sipping@ietf.org
Cc: victor@tamura-broad.com
Subject: Re: [Sipping] I-D ACTION:draft-victor-fullmesh-reqts-00.txt
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On Friday, May 30 2003, "Internet-Drafts@ietf.org" wrote to "IETF-Announce: ;, sipping@ietf.org" saying:

> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> 
> 
> 	Title		: Requirements for Full-mesh Conference Model using 
>                           Baseline Session Initiation Protocol
> 	Author(s)	: V. Paulsamy
> 	Filename	: draft-victor-fullmesh-reqts-00.txt
> 	Pages		: 7
> 	Date		: 2003-5-29
> 	
> This memo enumerates requirements to realize full-mesh multiparty 
> conferencing using baseline Session Initiation Protocol. Primary 
> requirement is to advertise conference membersĘ addresses to new 
> participants by concatenating multiple SDP session descriptions 
> together. This particular requirement can solve several well-known 
> problems, including disjoint meshes and race conditions, that are 
> lingering in full-mesh conferencing for a while now.

I've read over this document; it's not really a requirements document, but
rather an argument for a particular approach to solving the problem.

It seems to imply that a minor protocol modification can be used to cause
baseline SIP to support full mesh conferences.  This clearly isn't true; end
points are going to have to negotiate support for it, since current end
points don't support it.  The idea of carrying end point identifiers in SDP
isn't necessarily a bad one, though.

I've been studying on full mesh conferencing, and after writing a simulator
for full mesh conferencing, using a simplified model of SIP, I think the
problem is rather more complicated than Victor realizes.

I've written a paper on it, which I'll be presenting at NOSSDAV next week.
It's available from my homepage --
<http://www1.cs.columbia.edu/~lennox/fullmesh.pdf>.  I think this would be
useful input to further discussion of full mesh conferencing, since I've
explained a lot of the relevant issues there rather better than I could do
it off the top of my head in e-mail.

-- 
Jonathan Lennox
lennox@cs.columbia.edu
 
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Reply-To: <victor@tamura-broad.com>
From: "Victor A Paulsamy" <victor@tamura-broad.com>
To: "'Jonathan Lennox'" <lennox@cs.columbia.edu>, <sipping@ietf.org>
Cc: <victor@tamura-broad.com>
Subject: RE: [Sipping] I-D ACTION:draft-victor-fullmesh-reqts-00.txt
Date: Fri, 30 May 2003 16:31:54 -0700
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Inline ...

-----Original Message-----
>From: Jonathan Lennox [mailto:lennox@cs.columbia.edu] 

>It seems to imply that a minor protocol modification can be used to cause
>baseline SIP to support full mesh conferences.  This clearly isn't true;
>end
>points are going to have to negotiate support for it, since current end
>points don't support it.  The idea of carrying end point identifiers in SDP
>isn't necessarily a bad one, though.

This is a good point; I overlooked it. I'm inclined to solving it using a,
"session-level property attribute that applies to conference as a whole"
[from 2327]. It could be something like "a=playoutmix." This attribute needs
to be defined.

When "a=playoutmix" is present, it indicates that an end system supports
session-level mixing for play out and if absent, it is not supported. Hence,
to support full-mesh conferencing, both offer and answer must have
"a=playoutmix" attribute. 

When inviting a new entity into the conference, the offerer carries
session-level "a=playoutmix." The answer must contain "a=playoutmix" to
establish a dialog. Otherwise, it's ACKed and BYE is sent to terminate the
dialog (as usual). 

In the case of new ones joining the conference, if the incoming request
didn't have "a=playoutmix", a member can choose to discard it, silently.

Regards,

--victor

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        "'sipping@ietf.org'" <sipping@ietf.org>,
        "'Henning Schulzrinne'" <hgs@cs.columbia.edu>
Subject: Re: [Sipping] I-D ACTION:draft-elwell-sipping-identity-interworki
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Francois Audet wrote:
> Hi John,
> 
> In the SIP BNF description in RFC3261, on page 223, it is stated
> that if the userinfo in the SIP-URI is a telephone-subscriber
> (as opposed to a user), then the telephone-subscriber BNF is
> defined in RFC2806. (Although not explicitely stated, I assume
> it also means that the "user=phone" user-param needs to be
> included). Perhaps Henning can confirm?

Yes.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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Subject: Re: [Sipping] SIP and TCP transport....
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Sarkar, Uttam wrote:
> Hi All,
> I have posted this messgage to the sip-implementors list. There might have
> been a discussion already regarding this.
> When transport is TCP, do you keep the socket connection open till the call
> is active or you close the socket once party A and party B are connected.

The specification only requires that you keep a connection open until 
there have been 64*T1 seconds of inactivity. However, genreally 
speaking it is good to keep the connections open for as long as you 
have resources, since they could potentially be reused down the road.

> When you get BYE or other sip Message from either end, then you open new
> socket connection to the other side to send that request. I would appreciate
> your inputs.

Genreally there is a separate connection used for requests in each 
direction.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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From: "simon issa" <simon_issa79@hotmail.com>
To: sipping@ietf.org
Date: Sat, 31 May 2003 11:49:07 +0000
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Subject: [Sipping] Simple
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Dear Sirs,

As a BS telecommunication senior I have my project based on SIP/SIMPLE and I 
would like to ask some questions.

To your knowledge , is there any SIMPLE library , because I need SIMPLE 
library in my project not all the SIP library.
To your knowledge also , is there a good descriptive sip library , 
describing the use and the goal of the functions, also theirs arguments , is 
there any examples (programs) of the use of that library ?
any help would be appreciated.

Thanks in advance .

Regards.

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