[Iptel] RE: [Sipping] New draft on end system communication services

Xiaotao Wu <xiaotaow@cs.columbia.edu> Fri, 18 February 2005 14:45 UTC

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Date: Fri, 18 Feb 2005 09:20:04 -0500
From: Xiaotao Wu <xiaotaow@cs.columbia.edu>
To: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
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Cc: iptel@ietf.org, Toip list <toip@snowshore.com>, sipping@ietf.org, "Henning G. Schulzrinne" <hgs@cs.columbia.edu>
Subject: [Iptel] RE: [Sipping] New draft on end system communication services
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On Fri, 18 Feb 2005, Gunnar Hellstrom wrote:

> I found this to be a good draft.
>
> We have a general agreement in the Sipping group that there are three real time media that
> must be considered in all specifications. It is video, text and audio.

Gunnar,

Thank you for the comments.
Yes, I will put 'text' to the specification.
The 'text' media type will be for a session based text conversation, like
using MSRP. SIP MESSAGE method will be handled in Instant Messaging
extension.

-Xiaotao

>
> In the Less specification, video and audio are mentioned, but not text.
> Here are proposals for adding real time text to this specification.
>
> I would appreciate if other authors also follow this agreement.
>
>
> ---------------------------------------------------------------------------
> In section 10.1.1, there is:
>
> Parameters:  "media"        A list of medias. Possible values are "audio",
>                                 and "video".
>
> Proposal, I suggest to extend it to:
>
> Parameters:  "media"        A list of medias. Possible values are "audio",
>                                 "text" and "video".
>
> ------------------------------------------------------------------------------------
> Same section 10.1.1, next parameter
>
>                  "input"        The source of input. It can be "microphone",
>                                 for audio, "camera" for video, or a URL
>                                 to a media file.
>
> I think this parameter may need quite a bit of extension, such as "line" as an alternative
> for audio input as well as URL, "line" for video etc. I would expect that you also would
> like to control output, such as headset or loudspeaker for video, screen or speech
> synthesis for text etc. I leave that for further discussion, when the requirements are
> more clear,
>  but just request that you start with adding "keyboard" for text input.
>
> Proposal:
>
>                  "input"        The source of input. It can be "microphone",
>                                 for audio, "keyboard" for text, "camera" for video,
> 					  or a URL to a media file.
> ----------------------------------------------------------------------------------
> In section 11.1.2 text should be mentioned among the media.
>
> 11.1.2 "Midcall:merge" Action
>
>     "Midcall:merge" actions will merge multiple calls into an end system
>     hosted conference. By default, audio streams will be mixed and sent
>     to all the call participants. Video streams are forwarded to all the
>     call participants. If there are held or muted calls, the hold and
>     mute status are kept.  Their syntax is shown in Figure 14.
>
> Proposal:
>
> 11.1.2 "Midcall:merge" Action
>
>     "Midcall:merge" actions will merge multiple calls into an end system
>     hosted conference. By default, audio streams will be mixed and sent
>     to all the call participants. Video and text streams are forwarded
>     to all the call participants. If there are held or muted calls, the
>     hold and mute status are kept.  Their syntax is shown in Figure 14.
> ---------------------------------------------------------------------------------
>
> Regards
>
> Gunnar
>
> -------------------------------------------
> Gunnar Hellstrom
> Omnitor AB
> SWEDEN
> Mob: +46 708 204 288
> www.omnitor.se
> Gunnar.Hellstrom@Omnitor.se
> --------------------------------------------
>
>
> >-----Original Message-----
> >From: sipping-bounces@ietf.org [mailto:sipping-bounces@ietf.org]On
> >Behalf Of Xiaotao Wu
> >Sent: Friday, February 18, 2005 1:33 AM
> >To: sipping@ietf.org; iptel@ietf.org
> >Cc: Henning G. Schulzrinne
> >Subject: [Sipping] New draft on end system communication services
> >
> >
> >We have written a new draft on the Language for End System Services (LESS)
> >
> >http://www.ietf.org/internet-drafts/draft-wu-iptel-less-00.txt
> >
> >This work can be closely related to the SIP P2P work because in a P2P
> >network, there is no infra-structure to provide communication services,
> >for example, phone spam filtering. Usually, users will not put their
> >services on peers' node due to privacy and security reasons. That makes
> >end system services very important, and it is necessary to define a
> >mechanism for end system service creation.
> >
> >>From the abstract:
> >
> >    In Internet telephony, end systems can take a large role in providing
> >    services, especially in networks without pre-configured
> >    infra-structure, such as peer-to-peer networks.  Since we believe
> >    that end system services differ in their requirements from network
> >    services, we define a new service creation scripting language called
> >    the Language for End System Services (LESS).  LESS inherits many
> >    characteristics from the Call Processing Language (CPL).  It contains
> >    commands and events for direct user interaction and the control of
> >    media applications.  This document defines the basic elements of LESS
> >    and several commonly used LESS extensions.
> >
> >Suggestions and comments would be appreciated.
> >
> >-Xiaotao
> >
> >===========================================================
> >Name      : Xiaotao Wu
> >Email     : xiaotaow@cs.columbia.edu
> >Homepage  : http://www.cs.columbia.edu/~xiaotaow
> >Phone     : (212)939-7054,  Fax: (212)666-0140
> >Phone-PC  : (212)939-7133
> >SIP       : sip:xiaotaow@conductor.cs.columbia.edu
> >Office    : Room 506, Computer Science building, West 120th
> >===========================================================
> >
> >_______________________________________________
> >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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