From rem-conf Sat Jan 02 07:59:07 1999 
From rem-conf-request@es.net Sat Jan 02 07:59:06 1999
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To: <trouble@es.net>
From: <tip_iceberg@writeme.com>
Subject: Free Tip
Date: ש, 2 ינואר 1999 14:09:43
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Dear Investor / Business person,

Free Tip Iceberg would like you to have the following tip on world 
currency:

Dollar is going to go up vs. the Swiss Franc between 5% and 8% in the 
next few weeks.

We hope you find it useful.

And remember, this is just the Tip of the Iceberg.

regards,
Free Tip Iceberg team.

note: if you are not an investor or business person or otherwise do not 
want to get a free tip, please reply with subject "remove".
 
 
 
 
 
 
 
 
 
 
 



From rem-conf Sun Jan 03 18:27:52 1999 
From rem-conf-request@es.net Sun Jan 03 18:27:51 1999
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From: sozni@listbot.com
To: sozni@listbot.com
Subject: Ordinary mom's & grandma's, free nude pic's!
Message-Id: <E0zwzYE-0005Fd-00@mail1.es.net>
Date: Sun, 3 Jan 1999 18:15:26 -0800
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MOM's and GRANDMA's: http://www.teenfree.com/grandmalover

Amateur women's, ALL over 40 (mom's, grandma's, wife's, neighboor's, secretary's, girlfriend's, etc...) waiting you wet. 
FREE PIC's by category: 
Straight, Double Fucking, Sucking, Lesbo, Nude on shower, Pissing, Anal, Big ass, Group SEX, Dildo's and a Special VISITORS GALLERY.
AND MORE :

sozni - http://www.biosys.net/sozni




From rem-conf Mon Jan 04 00:12:56 1999 
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Reply-To: <netsurf@sersol.com>
From: "James D. Wilson" <netsurf@sersol.com>
To: "Abuse@Ftc. Gov" <abuse@ftc.gov>, <abuse@nkn.net>, <pathos@nkn.net>,
        <postmaster@nkn.net>, <techsup@nkn.net>, <abuse@bellatlantic.net>,
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Subject: FW: Ordinary mom's & grandma's, free nude pic's!
Date: Sun, 3 Jan 1999 21:57:30 -1000
Message-ID: <002a01be37b7$e119cd60$e5ff7fce@www.glhawaii.org>
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1


US Federal Trade Commission:  documentation of interstate theft of
electronic resources - please prosecute

NKN.EDU/NKN.NET:  These spammers are on your wire/are spamming on your
watch and using websites on your network to collect the profits of
their network abuse, network trespassing, and network theft of
services.  Please NUKE them to 5,000,000 deg. kelvin.  

Bellatlantic:  they used your wires to steal their services.  Please
nuke.

Biosys.net:  the spammer is spamvertizing a webpage on your network. 
Please NUKE immediately.  No honest business does business with
thieves.

Myriad.net:  please ensure that Biosys nukes the crooks off the net.

Listbot.com:  they are using your email services - please nuke sozni
the scumbag.

Netforward.com:  please dump the spammers.

National Knowledge Networks, Inc. (NETBLK-NKN-BLK-1) NKN-BLK-1
						 207.55.128.0 - 207.55.223.255
Abid Mustafa (NETBLK-NKN-ADEA1)	NKN-ADEA1      207.55.136.128 -
207.55.136.135

To single out one record, look it up with "!xxx", where xxx is the
handle, shown in parenthesis following the name, which comes first.

The ARIN Registration Services Host contains ONLY Internet
Network Information: Networks, ASN's, and related POC's.
Please use the whois server at rs.internic.net for DOMAIN related
Information and nic.mil for NIPRNET Information.

**complete**

National Knowledge Networks, Inc. (NETBLK-NKN-BLK-1)
   350 N. St. Paul - Suite 2410
   Dallas, TX 75205
   US

   Netname: NKN-BLK-1
   Netblock: 207.55.128.0 - 207.55.223.255
   Maintainer: NKN

   Coordinator:
      Pathos, Peter  (PP46-ARIN)  pathos@NKN.NET
      214-880-0702

   Domain System inverse mapping provided by:

   DFW.NKN.EDU			199.171.20.1
   NS2.NKN.NET			199.171.20.31

   Record last updated on 17-Nov-97.
   Database last updated on 1-Jan-99 16:11:45 EDT.

The ARIN Registration Services Host contains ONLY Internet
Network Information: Networks, ASN's, and related POC's.
Please use the whois server at rs.internic.net for DOMAIN related
Information and nic.mil for NIPRNET Information.

**complete**

Abid Mustafa (NETBLK-NKN-ADEA1)
   3951 SW Gamwell
   Topeka, KS 66610
   US

   Netname: NKN-ADEA1
   Netblock: 207.55.136.128 - 207.55.136.135

   Coordinator:
      Technical Support  (TS135-ORG-ARIN)  techsup@NKN.NET
      214-954-0028
Fax- 214-740-1872

   Record last updated on 29-Oct-97.
   Database last updated on 1-Jan-99 16:11:45 EDT.

The ARIN Registration Services Host contains ONLY Internet
Network Information: Networks, ASN's, and related POC's.
Please use the whois server at rs.internic.net for DOMAIN related
Information and nic.mil for NIPRNET Information.

**complete**

Host info www.biosys.net 2/3000/50/0; 01/03/99 21:52:10

Official name: www.biosys.net
IP address: 204.57.67.39

ping 204.57.67.39       packet 2            failed, retcode = 11010
(Timed Out)


Registrant:
BioSys.Net Services (BIOSYS2-DOM)
   700 University Drive East, Suite 106
   College Station, TX 77840
   US

   Domain Name: BIOSYS.NET

   Administrative Contact, Technical Contact, Zone Contact:
      NetForward Admin  (NA174-ORG)  admin@NETFORWARD.COM
      (409)693-8885
Fax- (409)693-6923
   Billing Contact:
      NetForward Admin  (NA174-ORG)  admin@NETFORWARD.COM
      (409)693-8885
Fax- (409)693-6923

   Record last updated on 08-Jun-97.
   Record created on 07-Mar-97.
   Database last updated on 3-Jan-99 17:57:14 EST.

   Domain servers in listed order:

   NS.MYRIAD.NET		204.57.67.2
   NS2.MYRIAD.NET		204.57.67.20

- -
James D. Wilson

"non sunt multiplicanda entia praeter necessitatem"
    William of Ockham (1285-1347/49)
 

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From: sozni@listbot.com
To: sozni@listbot.com
Subject: Ordinary mom's & grandma's, free nude pic's!
Message-Id: <E0zwzYE-0005Fd-00@mail1.es.net>
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Sent: Sunday, January 03, 1999 4:15 PM
To: sozni@listbot.com
Subject: Ordinary mom's & grandma's, free nude pic's!


MOM's and GRANDMA's: http://www.teenfree.com/grandmalover

Amateur women's, ALL over 40 (mom's, grandma's, wife's, neighboor's,
secretary's, girlfriend's, etc...) waiting you wet. 
FREE PIC's by category: 
Straight, Double Fucking, Sucking, Lesbo, Nude on shower, Pissing,
Anal, Big ass, Group SEX, Dildo's and a Special VISITORS GALLERY.
AND MORE :

sozni - http://www.biosys.net/sozni



-----BEGIN PGP SIGNATURE-----
Version: PGP 6.0.2
Comment: Spammers are NetAbusers - Jail Them With The Other Criminals

iQA/AwUBNpB0ajAufbtGOmgdEQJ7PACeNEGv6/WbJRIXt3hJ3jK7BeFgiC4AoJI2
SrObcjcR8dvdJlxdNB6HQW6K
=mvsa
-----END PGP SIGNATURE-----




From rem-conf Mon Jan 04 04:38:56 1999 
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Subject: RTP/UDP/IP header compression
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Hi,

In the Internet Draft - IP Header Compression over PPP,
assigned numbers for RTP/UDP/IP header compression packets are
given (COMPRESSED_RTP, COMPRESSED_UDP, CONTEXT_STATE, etc) . 

Could any of you tell me what are the assigned numbers for rfc1294
encapsulation over Frame Relay ? 

What about X25 link ?

Expecting your reply

Rgds

GVJohn

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Hi,

In the Internet Draft - IP Header Compression over PPP,
assigned numbers for RTP/UDP/IP header compression packets are
given . 

Can one of you tell me what are assigned numbers for rfc1294
encapsulation over Frame Relay ?

What about X25 link ?
Does it hold the same value as FR.

Expecting your reply

Rgds

GVJohn

--------------63FAD7033B4
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Date: Fri, 20 Nov 1998 09:44:16 +0530
From: Sriharsha J <harsha@miel.mot.com>
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Date: Thu, 19 Nov 1998 09:33:26 -0800 (Pacific Standard Time)
From: Stephen Casner <casner@cisco.com>
To: Sriharsha J <harsha@miel.mot.com>
Cc: van@ee.lbl.gov
Subject: Re: RTP/UDP/IP header compression
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> 1. What is the initial sequence number set to ? (zero ?)

Doesn't matter since the FULL_HEADER packet will cause that sequence
number, whatever it is, to be stored in the decompressor's context.
Zero is fine.

> 2. When the synchronization is lost, what does the sequence # field of
>     the CONTEXT_STATE packet set to ?
>     If you have received packet (X-1) and expecting (X), you got (X+k).
>     Then do you send (X-1 or X) ?

It should be the last valid sequence number, i.e., X-1.  You are right
that this point is not clear enough in the first paragraph of 3.3.5,
although it might be inferred from the second paragraph on advisory
CONTEXT_STATE packets.  In a sense, again it doesn't matter because if
the I bit is set to 1, the compressor must send a FULL_HEADER packet
regardless of what the sequence number in the CONTEXT_STATE packet is,
and the FULL_HEADER packet will cause the sequence number in the
decompressor to be set explicitly.  The new value that it gets set to
will usually NOT be X because the compressor does not buffer packets.

> 3. How does the Compressor detect duplicate CONTEXT_STATE packets ?
>     (through sequence # ?)

They are duplicate if they are identical; that is, yes, if the
sequence number is the same.  Clearly, the same sequence number will
be used again after 16 more data packets, so the compressor must take
timing into account.  If there are 16 packets in flight on the link,
the compressor might not be sure about which packet was indicated.
But that represents such a long delay that the performance of this
protocol on such a link is not likely to be acceptable, so it should
not be used.

> In the above context, does generation number has any role or not ?

The generation number has no role.
							-- Steve



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--------------63FAD7033B4--



--------------243D1A546F8--




From rem-conf Mon Jan 04 06:01:58 1999 
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To: itc@ieee.org, comsoc-chapters@ieee.org, multicomm@cc.bellcore.com,
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Subject: Announcement for ICNP'99, in Toronto, Canada
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			   CALL FOR PAPERS

	7th IEEE INTERNATIONAL CONFERENCE on NETWORK PROTOCOLS

		           Toronto, Canada

		    October 31 - November 3, 1999

	     http://www.cs.columbia.edu/~hgs/edas/icnp99/

In just 6 years ICNP has established itself as one of the premier
conferences in the computer networking field. ICNP deals with all
aspects of communication protocols, from design and specification, to
verification, testing, performance analysis, implementation, and
performance tuning.  Protocol functions of interest include (but are
not limited to) network access, switching, routing, flow and
congestion control, multimedia transport, wireless protocols, support
for mobility, multicast, network security, Web protocols and
applications, electronic commerce, network management,
interoperability, and internetworking.

ICNP uses a single-track format which provides an intimate setting for
discussion and debate. The conference is known for its high quality
papers from the research communities of IEEE COMSOC, IEEE Computer
Society, and ACM SIGCOMM.  To be considered for acceptance, a
submission should present a significant contribution in its topic
area. Selected papers will be forwarded for "fast track" consideration
to IEEE/ACM Transactions on Networking. ICNP also includes panel
sessions in which experts in a specific area offer their observations
and opinions about a current topic.

ICNP'99 will be held in Toronto, whose name is a Huron Indian word
meaning "place of meeting".  Toronto is Canada's largest city, the
capital of the province of Ontario, and one of the most exciting,
vibrant, and progressive cities in the world! It's attractions are far
too numerous to list. The United Nations has designated Toronto as the
world's "most ethnically-diverse city".  It has the world's tallest
building, the CN Tower, is the third-largest theatre centre in the
English-speaking world, and contains over 5,000 restaurants and
eateries. Toronto is also a very safe city with many family
attractions.


Important Dates
---------------

Paper Submission Deadline:  	May 3, 1999
Notification of Acceptance:  	August 2, 1999

Camera Ready Due:  		August 25, 1999

Tutorials:  			October 31, 1999
Conference:			November 1-3, 1999


Submission Information
----------------------

Submissions will be made via email. Details of the procedure are given
at the ICNP'99 web site:

http://www.cs.columbia.edu/~hgs/edas/icnp99/


General Chair
-------------
Johnny Wong, University of Waterloo
jwwong@bcr.uwaterloo.ca


Program Chairs
--------------
Joe Bannister, USC Information Sciences Institute
Email: joseph@isi.edu

Terry Todd, McMaster University
Email: todd@mcmaster.ca


Tutorial Chair
--------------
Kevin Almeroth, UC Santa Barbara
Email: almeroth@cs.ucsb.edu


Local Arrangements Chair
------------------------
Bart Domzy, Trent University


ICNP Steering Committee
-----------------------
Mostafa Ammar, Georgia Insitute of Technology
Mohamed Gouda, University of Texas at Austin
Simon Lam, University of Texas at Austin
David Lee, Bell Labs
Ming T. (Mike) Liu, Ohio State University
Raymond Miller, University of Maryland, College Park
Krishan Sabnani, Bell Labs

ICNP is sponsored by the IEEE Computer Society








From rem-conf Mon Jan 04 09:33:41 1999 
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To: SKYSHOP OFFICE SUPPLIES <skyshop@excite.com>
cc: rem-conf <rem-conf@es.net>, P.Kirstein@cs.ucl.ac.uk
Subject: SP
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In message <419.436158.61900648skyshop@excite.com>you write:
>Are you paying to much for office supplies? Skyshop CAN save you money.  
>
>If you would like to see our product line & website please email us back in th
>e subject 
>line of the below email address and place the letters (SP)
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>                skyshop@gte.net
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Peter

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From rem-conf Mon Jan 04 14:46:52 1999 
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Date: Mon, 4 Jan 1999 14:42:54 -0800 (Pacific Standard Time)
From: Stephen Casner <casner@cisco.com>
To: Geevarghese John <gvjohn@miel.mot.com>
cc: van@ee.lbl.gov, rem-conf@es.net
Subject: Re: RTP/UDP/IP header compression
In-Reply-To: <3690B2D0.75D6@miel.mot.com>
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On Mon, 4 Jan 1999, Geevarghese John wrote:

> In the Internet Draft - IP Header Compression over PPP,
> assigned numbers for RTP/UDP/IP header compression packets are
> given (COMPRESSED_RTP, COMPRESSED_UDP, CONTEXT_STATE, etc) . 
> 
> Could any of you tell me what are the assigned numbers for rfc1294
> encapsulation over Frame Relay ? 
> 
> What about X25 link ?

To my knowledge, no assignments have been made for Frame Relay or X25
link layers.  Some link layers may inherit assignments from PPP, but
for those that don't, it will be necessary to get assignments from the
appropriate authorities in order to use CRTP.

If it is not feasible to get multiple assignments for some link layer,
one could define a multiplexing octet along the lines of what is in
Section 13 of draft-degermark-ipv6-hc-08.txt.
							-- Steve




From rem-conf Tue Jan 05 00:23:48 1999 
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To: Yozo Toda <yozo@aohakobe.ipc.chiba-u.ac.jp>
cc: rem-conf@es.net
Subject: Re: vic-2.8 problem. 
In-reply-to: Your message of "Tue, 22 Dec 98 19:09:40 PST."
             <199812230309.MAA16939@aohakobe.ipc.chiba-u.ac.jp> 
Date: Tue, 5 Jan 1999 00:11:05 PST
Sender: Bill Fenner <fenner@parc.xerox.com>
From: Bill Fenner <fenner@parc.xerox.com>
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There are several memory leaks in vic-2.8 and vat-4.0 .

The patches in the FreeBSD port system
(http://www.freebsd.org/ports/mbone.html) fix the memory leaks and
also allow building with tcl/tk 8.0 .  I don't know if anyone is
planning to do any more vic or vat releases.

  Bill



From rem-conf Tue Jan 05 04:21:07 1999 
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To: rem-conf@es.net
Subject: Draft AVT minutes
cc: casner@cisco.com
Date: Tue, 05 Jan 1999 12:12:58 +0100
Message-ID: <1304.915538378@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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The draft minutes from the AVT working group meeting in Orlando are
enclosed. Due to the imminent submission deadline, I'd appreciate any
comments on these as soon as possible. 

Colin

-------------------------------------------------------------------------------
Minutes of the Audio/Video Transport Working Group

Reported by Colin Perkins.

The audio/video transport working group met twice at the 43rd IETF in
Orlando. The main agenda items were discussion of RTP multiplexing, and the
update of RTP and the audio/video profile for advancement to draft standard
status. In addition, a number of RTP payload formats and the RTP MIB were
discussed.

The meeting opened with a review of the working group status, and documents
completed since the last meeting. These include the payload formats for
H.263+ video (RFC2429), BT.656 video (RFC2431) and JPEG video (RFC2435),
together with AT&T's Error Resilient Video Transmission Technique (RFC2448).

The RTP MIB (draft-ietf-avt-rtp-mib-03.txt) was presented by Mark Baugher.
Changes since the previous draft have been motivated by comments and
careful review by Bert Wijnen and the ITU SG16. The main differences are as
follows: renamed MIB definitions to RTP-MB and module to RTPMIB; changed
OID structure to current IETF conventions; several clarifications made to
DESCRIPTION clauses; use "noSuchInstance" rather than "noSuchObject" in
rtpRcvrRTT; changed media counters to be 64 bits; separated host and
monitor compliance sections; and the use InterfaceIndexOrZero instead of
redefining InterfaceIndex.

There are a number of open issues with the use of multiple MIB agents in
switches, the optional nature of the interface index and to permit easier
selection of session entries by session address which have to be resolved,
but it is expected that this will be complete within the next few weeks, at
which time a new draft will be issued ready for working group last call. A
reference implementation is available of the -02 draft, work is progressing
on the latest version. The RTP MIB is now referenced in the ITU as H.341.

The RTP payload format for DTMF and other telephone tones
(draft-ietf-avt-dtmf-01.txt, draft-ietf-avt- tones-00.txt) was presented by
Scott Petrack. The justification for using a new payload format is to
reproduce the tones better than current, low rate, codecs; to apply
redundancy differently for tones and speech; to separate detection of tones
>from their interpretation; and to preserve the sounds associated with
particular signals. Two payload formats are defined: a named signal event
payload and a tone frequency format payload. 

The named signal event payload includes a named event (dialtone, busy,
call-waiting, off-hook, etc), together with the volume and duration of that
event. The tone frequency format sends a representation of the actual set
of tones to be played, rather than either purpose (name), enabling a wider
range of tones and the playout of foreign tones. 

Both formats may be conveyed in a single packet, using RFC2198 redundancy,
for robustness. Mark Handley expressed some concern with the means by which
this is done: in particular it is unclear how to playout these tones when a
fraction of the tone is lost. 

The two drafts are to be merged within the next couple of months, and it is
the desire of the authors to move the result to the standards track soon
after. Some concern was expressed that this draft is not receiving
sufficient exposure in bodies such as the ITU which may be able to provide
relevant feedback - it must be ensured that these bodies have chance to
comment before this document can progress in the IETF.

The discussion of RTP multiplexing started with a presentation by Jonathan
Rosenberg on issues in RTP multiplexing (draft-ietf-avt-muxissues-00.txt).
As noted in this draft, there are two scenarios which are of interest:
end-to-end and mid-network multiplexing. The scenario chosen will affect
the choices made in designing the multiplexing scheme. These choices affect
the delineation, identification, synchronisation and dynamism of the
multiplexed data.

There are three means by which data in a multiplexed packet may be
delineated: by explicit length indicators (which have maximum flexibility
but the largest overhead); implicitly based on payload type (many codecs
are fixed length blocks); or implicitly by out of band signalling (which
has the least bandwidth overhead, but requires signalling and makes
changing encodings or packetisation difficult).   If all multiplexed
packets are of the same duration this problem becomes much simpler.

Multiplexed data may be identified by an explicit ID, where the number of
bits used depends on the number of simultaneous calls and the desired ID
reuse latency. This ID may be the RTP SSRC in some cases. Alternatively the
data may be channelised, where each user's data gets a slot in the packet.
This latter approach requires out-of-band signalling and, of silence
suppression is used, a bitmask to indicate which channels are active. The
overheads of the two schemes vary with the number of channels being
multiplexed, whether channels are active continually, and with the rate of
change of the set of multiplexed streams.

If each frame of multiplexed data within a packet has the same timestamp,
the individual timestamps may be elided, and replaced by the single
timestamp in the multiplexed packet's RTP header. If the timestamps are
close, offsets relative to the outer timestamp may be used rather than
complete timestamps (saving some number of bits). Alternatively if users
have uncorrelated frame start times it is necessary to send the complete
timestamp per user.

The dynamism of codecs also affects the multiplexing format chosen. If
codecs never change there is no need to include payload type indication
within the multiplexed stream, and out of band signalling may be used.
Similarly, of codecs change rarely out of band signalling may be
appropriate, depending on how often codecs may change (synchronisation
between the signalling of a payload type change and the media stream may be
complex). If codecs change frequently, some form of in-band payload type
indication is most appropriate - this need not necessarily be a complete
RTP PT value if the set of allowable codecs is small a mapping table may be
used instead.

The marker bit may need to be transmitted depending on the use of the
multiplexed stream.

It was noted that this is a very large space, and a number of solutions to
the multiplexing problem are possible. This group has a number of solutions
presented to it, yet the precise problem definition for each of these has
not been enunciated. It may help to focus the discussion if the question
"why are you multiplexing?" is clearly answered, and if we derive a number
of scenarios which require common solutions.

Essentially, we need to focus on requirements. trying to do generic
optimisation using a multiplexer is futile.

It was noted that there is some RTP compression and transport work being
conducted in the ATM forum which may be relevant to this discussion.

The first multiplexing proposal (an update to draft-ietf-avt-mux-rtp-00.txt) 
was presented by Barani Subbiah. The stated goals of this proposal are to
achieve the best possible fit with cellular/PSTN applications and to derive
a payload format suitable for use in a switched IP telephone network. They
are aiming for a simple format with a fixed header suitable for hardware
implementation, providing a compromise between bandwidth saving (in
addition to the outer RTP header, this proposal averages two bytes overhead
per multiplexed stream) and complexity.

The sequence number (2 bits) is new since the last draft, this means a
reduction in the length field size to 5 bits. Concern was expressed that 
a 5 bit length field is insufficient for some audio codecs which may be
desirable. The use of the 2 bit sequence number was questioned, since 4
packet losses are possible -- a longer field should probably be used.

A transition bit is included to signal a change in the end-to-end flow
parameters, allowing one state change per RTT. It was noted that if all
packets with an RTT are lost, the state change will not be noted at the
receiver. IT was also noted that payloads such as DTMF interspersed with
voice can cause change in payload type more often than once per RTT.

Tohru Hoshi presented draft-tanigawa-rtp-multiplex-01.txt. This is a simple
proposal where multiple RTP packets are concatenated for transmission in a
single UDP packet. The default packetisation interval specified for a codec
in the audio/video profile is used such that no length indication is
necessary. The draft now includes a section describing the efficiency gains
for using this proposal according to various metrics. Call setup signalling
is also defined.

The next multiplexing proposal discussed was GeRM (draft-ietf-avt-germ-00.txt),  
presented by Mark Handley. The goal of this proposal is to transparantely
multiplex a number of RTP streams. It operates using difference coding
between the headers of packets to be multiplexed together. Clearly this
will work better if the packet headers are similar (this can be achieved
between co-operating gateways, although the traffic pattern will affect
performance) but it will still work if the end points do no co- operate,
and will perform no worse than simply concatenating packets.

The GeRM protocol is well suited for scenarios where a mix of RTP packets
are to be multiplexed, such as may be encountered in the transport of
MPEG-4 streams, or for use between a pair of cooperating gateways
multiplexing a large number of similar streams. It achieves considerable
flexibility, at the expense of complex parsing and greater bandwidth
overhead than other, less general, protocols. 

The final multiplexing presentation was by Dean Willis (no draft exists).
The assumptions here are that large numbers of streams are being carried
between end-point pairs, fast interfaces with minimal serialisation delay
are used, and mixed codecs with silence suppression exist. The goal is to
increase overall "network efficiency" by re-packing packets to increase the
total MTU and reduce the number of packets sent. Two alternatives were
considered:
- RTP level: complex, no benefit to non-RTP traffic, issues with RTCP
- UDP level: brute force level simplicity, aggregate UDP flows between
  end-points, allows IPsec at multiplexing level, transparent to the
  application, allows mid-network multiplexing, less efficient
The UDP level approach is favoured.

Following the presentations a considerable amount of discussion ensued.
Concern was expressed that multiplexing is being used as an "RTP switching"
solution, with application level routing: it was noted that IP has a number
of perfectly reasonable routing algorithms already, and it is unnecessary
to re-invent these within RTP. Many people expressed concern that the
problem to be solved by multiplexing has not been clearly stated: is it to
reduce the number of packets sent? the number of bytes sent? to perform
application level RTP routing? etc. It is unclear that RTP multiplexing is
the correct solution here: a generic UDP multiplexing protocol (as in the
final proposal) may be more suitable in some cases.

Concern was expressed that multiplexing streams with different transport
level addressing into one is not clearly handled by these proposals. In
some cases, the SSRC is assumed to provide a unique stream ID, which is 
not necessarily the case across multiple streams. The handling of RTCP data
by a number of these proposals is also unclear.

Concern was expressed that these payload formats should be independent of
the signalling to be used. We may want to express signalling requirements,
but should not tie these protocols to a specific scheme.

Since it is unclear whether a single protocol can satisfy both aims, and
which of the five proposals currently submitted to the group will go
forwards, the group decided to conduct more simulation and analysis of the
proposals, in order to facilitate a fair comparison.

The next subject was the transport of MPEG-4 streams within RTP. A number
of AVT members participated in the MPEG meeting in Atlantic City in
October, leading to the formation of an ad-hoc group within MPEG to discuss
MPEG transport using IP. The work conducted in that group to date was
presented by Reha Civanlar. 

A number of alternatives for the transport of MPEG-4 streams in an IP
network were considered: 
	- directly on UDP
	- RTP followed by a full MPEG-4 SyncLayer packet
	- MPEG-4 SyncLayer packets mapped onto RTP packets
	- MPEG-4 elementary streams over RTP with natural payload formats
The preferred approach would be to use the latter approach, but since the
ES interface is not a normative part of MPEG-4 this is not possible. 

The approach chosen is, therefore, to map the MPEG-4 SyncLayer packets onto
RTP packets, such that the common pieces of the header reside in the RTP
header, with a small payload header providing the MPEG-4 specific features.
A single payload format is therefore used for MPEG-4 streams transported
within RTP, and the MPEG-4 model is maintained (although not the precise
packet format). In this approach, an RTP multiplexing scheme is needed to
fulfill the role of FlexMux in MPEG-4. The GeRM proposal seems to be a good
fit for this.

An internet draft detailing this work is in preparation. Those who wish to 
participate in this work are encouraged to join the ad-hoc group's mailing
list: send email to 4onIP-sys-request@fzi.de in order to subscribe.

Christine Guillemot presented an RTP generic payload with scalable and
flexible error recovery (draft-guillemot-genrtp-00.txt). This draft takes
a somewhat different view of the problem of transporting MPEG-4 content and
is based on carrying elementary streams in a generic manner. The motivation
for this is to transport many types of stream whilst avoiding having to
define a payload format for each and allowing finer control of error
correction with a set of different FEC mechanisms and the possibility of
grouping AUs in a single packet.

One of the aims of this work are to factorize the common features instead
of developing specific formats for each codec/type of elementary stream and
to be able to identify repeated data so that the network adaptive layer can
identify and remove this if desired. The adaption layer can add FEC to
entire packets or to portions of a stream within packets (adding redundancy
in a similar manner to RFC2198).

Concern was expressed that adding FEC to portions of a packet adds a lot of
extra complexity, and unless this FEC is much smaller than that which would
otherwise be present this complexity may not be justified.

Some concern was expressed that this document includes a number of payload
formats (redundancy, FEC, fragmentation and grouping) which may be better
separated. This clearly depends on the details of the stream which is being
packetised.

It is unclear that this format is suitable as a generic RTP payload, however
it may work well as a general purpose transport for MPEG-4 elementary streams.

Steve Casner described the changes made to the main RTP specification since
the last meeting. This is now stable, and unless major problems are found
is believed ready for last-call for draft standard. The changes since the
last meeting are described in draft-ietf-avt-rtp-new-02.txt and include:
	- SSRC sampling moved to separate draft (ietf-avt-rtpsample-01.txt)
	- Keep only unconditional reconsideration 
	- Add IANA considerations section added; no longer suggest experimental
  	  registration of values
	- Y2036 (in)consequences explained
	- convert to MUST, SHOULD, MAY
A plea was made for help checking:
	- Section 0: resolution of open issues
	- Section 6.2: RTCP transmission interval
	- Section 6.3: RTCP send and receive rules
	- Appendix A: does the code work?
	- Appendix B: changes from rfc1889
It was noted that the group must document "at least two independent and
inter-operable implementations from different code bases" of "all of the
options and features of the specification" in order to advance to draft
standard status (RFC2026). Colin Perkins volunteered to produce a draft
detailing those options and features as a checklist for vendors to check
compliance, and Jonathan Rosenberg volunteered to produce a draft detailing
tests for the timer reconsideration algorithms.

The changes to the RTP profile (draft-ietf-avt-profile-new-04.txt) since
the last meeting are less advanced: a clearer statement of the new policy
of no more static assignments, and the addition of change bars. It is
still necessary to complete the update with MUST, SHOULD, MAY, etc and to
add text to allow default of 5% RTP bandwidth to be overridden. 

The registration of RTP payload format names as MIME types is still not
complete: Philipe Hoscha volunteered to work on this, and to work out the
details of the process. It is hoped that this may just be a statement we
can put in the profile to specify how the registration is done, without
changing the MIME registration process, but this is not yet clear.  

Steve Casner presented work on some new RTCP SDES items for Peter Parnes,
who was unable to be present (draft-parnes-rtp-sdes-00.txt). These items
include nickname, homepage, personal image, active media, (suggested
earlier: organisation). 

Concern was expressed about the inclusion of URLs in an SDES packet which
may be sent to a large multicast group, since simultaneous retrieval of
these by many receivers can cause implosion problems. Some form of random
backoff algorithm must be specified (as is done in this draft).

Concern was expressed that we shouldn't try to turn RTCP into a general
data transfer mechanism. The consensus of the group was that additional
information to be carried in RTCP must be optional, and should not affect
the operation of the application, and that an RFC must be present defining
the new RTCP packets, and giving an applicability statement for each. No
additional packet types will be added to the main RTP specification.

The SSRC sampling algorithms (draft-ietf-avt-rtpsample-01.txt) were
presented by Jonathan Rosenberg. These have been moved out of the main 
RTP specification because of the IPR issues (Lucent patent on the binning
algorithm). The changes to this document are:
	- Uniformity of SSRC values usage: recommend hashing SSRC value,
	  because some broken implementations doesn't choose uniformly
	  distributed SSRC values
	- New section on performance of sampling in terms of coefficient
	  variation added
	- An explicit statement of the IPR issues and licensing terms needs
	  to be added
Comments are requested. 

The document giving guidelines for writers of RTP payload formats
(draft-ietf-avt-rtp-format-guidelines-01.txt) was noted as being
essentially complete. One more revision will be produced to include
the final round of comment, before last call for BCP status. Comments
are sought as soon as possible.

The payload format for PureVoice(TM) audio (draft-mckay-qcelp-01.txt) was
noted as being in working group last call still, pending resolution of a
number of open issues with the encryption of the payload data in a
non-standard manner.

The Generic FEC payload (draft-ietf-avt-fec-04.txt) was presented by
Jonathan Rosenberg. Changes include the removal of Reed-Solomon coding 
(to a separate draft) and mask extension. The examples and code have
been tested and bug-fixed, and the issues with encryption during key
changes have been resolved. This document is essentially ready for last
call as proposed standard - will do one more revision to get the MUST,
SHOULDs, etc sorted. Comments are sought.

The Reed-Solomon draft (draft-ietf-avt-reedsolomon-00.txt) was also
presented by Jonathan Rosenberg. Help is required with this: if anyone 
is interested in seeing this work progress they should contact Jonathan,
else the draft will not be updated.

A proposal to use the RFC2198 redundancy format as a transport for
interleaved audio (draft-ietf-avt-interleaving-00.txt) was presented by
Colin Perkins. This may eventually be merged into that document, although
this is at present undecided. Comments are sought.

The meeting concluded with a brief presentation on the proposed charter
revision. It was noted that this needs clarification regarding MPEG-4
payload formats, but is otherwise satisfactory. The revised charter
will be sent for IESG approval in the near future.
-------------------------------------------------------------------------------



From rem-conf Tue Jan 05 08:46:12 1999 
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From: Carlos Eduardo Silva =?iso-8859-1?Q?Mart=EDnez?= 
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I want to know what are the most important features that must be tested
on a videoconference system (sony 5100), and how do it.




From rem-conf Tue Jan 05 20:11:19 1999 
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when someone queries the AltaVista search engine for a keyword related to your
site's products or services, does your page appear in the top 10
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If you're listed but not within the first two or three pages of results,
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A top 10 ranking in a major search engine like
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Over 90% of the search engine traffic to most Web sites comes from
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"After we used AstaLavista, our online sales went from a few thousand to
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-One of our Customers (identity not reveled for security).

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 -- Thank you!         Now for the good information!
*******************************************************

Hi,
I was just searching on AltaVista with the key word(s) "video"
and I see your website does not come up in the  top 10 listings.

when someone queries the AltaVista search engine for a keyword related to your
site's products or services, does your page appear in the top 10
matches, or does your competition? 

If you're listed but not within the first two or three pages of results,
 you lose, no matter how many engines you submitted your site to. 

A top 10 ranking in a major search engine like
AltaVista will often generate more targeted traffic than an expensive
banner advertising campaign - and, a good search engine position is
like highly targeted advertising that is both FREE, and effective! 

Over 90% of the search engine traffic to most Web sites comes from
8 to 10 major search engines from there first THREE PAGES!
Therefore, it is VERY important that your site comes up in the top 10
results.

"After we used AstaLavista, our online sales went from a few thousand to
over $75 thousand in one month from the time we signed up with them, (AstaLavista)
we are now dominating the  searches people make for our products".
-One of our Customers (identity not reveled for security).

We can get your page top 10 GUARANTEED!!! NO BULL.
for more info about what we do see our website at:www.geocities.com/
Hi,
I was just searching on AltaVista with the key words "computer printers"
and I see your website does not come up in the  top 10 listings.

when someone queries the AltaVista search engine for a keyword related to your
site's products or services, does your page appear in the top 10
matches, or does your competition? 

If you're listed but not within the first two or three pages of results,
 you lose, no matter how many engines you submitted your site to. 

A top 10 ranking in a major search engine like
AltaVista will often generate more targeted traffic than an expensive
banner advertising campaign - and, a good search engine position is
like highly targeted advertising that is both FREE, and effective! 

Over 90% of the search engine traffic to most Web sites comes from
8 to 10 major search engines from there first THREE PAGES!
Therefore, it is VERY important that your site comes up in the top 10
results.

"After we used AstaLavista, our online sales went from a few thousand to
over $75 thousand in one month from the time we signed up with them, (AstaLavista)
we are now dominating the  searches people make for our products".
-One of our Customers (identity not reveled for security).

We can get your page top 10 GUARANTEED!!! 
Try us out you have nothing to lose!
for more info about what we do see our website at:

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From rem-conf Tue Jan 05 20:44:41 1999 
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If this was an error on our part PLEASE let us know by sending a reply to this 
email with GET ME OFF, in the subject line, we keep track of all removes we get and you will
not be mailed again.
 -- Thank you!         Now for the good information!
*******************************************************

Hi,
I was just searching on AltaVista with the key word(s) "video"
and I see your website does not come up in the  top 10 listings.

when someone queries the AltaVista search engine for a keyword related to your
site's products or services, does your page appear in the top 10
matches, or does your competition? 

If you're listed but not within the first two or three pages of results,
 you lose, no matter how many engines you submitted your site to. 

A top 10 ranking in a major search engine like
AltaVista will often generate more targeted traffic than an expensive
banner advertising campaign - and, a good search engine position is
like highly targeted advertising that is both FREE, and effective! 

Over 90% of the search engine traffic to most Web sites comes from
8 to 10 major search engines from there first THREE PAGES!
Therefore, it is VERY important that your site comes up in the top 10
results.

"After we used AstaLavista, our online sales went from a few thousand to
over $75 thousand in one month from the time we signed up with them, (AstaLavista)
we are now dominating the  searches people make for our products".
-One of our Customers (identity not reveled for security).

We can get your page top 10 GUARANTEED!!! NO BULL.
for more info about what we do see our website at:www.geocities.com/
Hi,
I was just searching on AltaVista with the key words "computer printers"
and I see your website does not come up in the  top 10 listings.

when someone queries the AltaVista search engine for a keyword related to your
site's products or services, does your page appear in the top 10
matches, or does your competition? 

If you're listed but not within the first two or three pages of results,
 you lose, no matter how many engines you submitted your site to. 

A top 10 ranking in a major search engine like
AltaVista will often generate more targeted traffic than an expensive
banner advertising campaign - and, a good search engine position is
like highly targeted advertising that is both FREE, and effective! 

Over 90% of the search engine traffic to most Web sites comes from
8 to 10 major search engines from there first THREE PAGES!
Therefore, it is VERY important that your site comes up in the top 10
results.

"After we used AstaLavista, our online sales went from a few thousand to
over $75 thousand in one month from the time we signed up with them, (AstaLavista)
we are now dominating the  searches people make for our products".
-One of our Customers (identity not reveled for security).

We can get your page top 10 GUARANTEED!!! 
Try us out you have nothing to lose!
for more info about what we do see our website at:

http://www.geocities.com/RodeoDrive/Outlet/4974/index.html
or email me at: obino@geocities.com






From rem-conf Wed Jan 06 02:32:34 1999 
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Enclosed is the final version of the AVT minutes from the Orlando IETF
meeting. Copies of the presentation slides and these minutes may also 
be found at http://www-mice.cs.ucl.ac.uk/multimedia/misc/avt/

Colin


-------------------------------------------------------------------------------
Minutes of the Audio/Video Transport Working Group

Reported by Colin Perkins.

The audio/video transport working group met twice at the 43rd IETF in
Orlando. The main agenda items were discussion of RTP multiplexing, and the
update of RTP and the audio/video profile for advancement to draft standard
status. In addition, a number of RTP payload formats and the RTP MIB were
discussed.

The meeting opened with a review of the working group status, and documents
completed since the last meeting. These include the payload formats for
H.263+ video (RFC2429), BT.656 video (RFC2431) and JPEG video (RFC2435),
together with AT&T's Error Resilient Video Transmission Technique (RFC2448).

The RTP MIB (draft-ietf-avt-rtp-mib-03.txt) was presented by Mark Baugher.
Changes since the previous draft have been motivated by comments and
review by Bert Wijnen and the ITU SG16. The main differences are as
follows: renamed MIB definitions to RTP-MIB and module to RTPMIB; changed
OID structure to current IETF conventions; several clarifications made to
DESCRIPTION clauses; use "noSuchInstance" rather than "noSuchObject" in
rtpRcvrRTT; changed media counters to be 64 bits; separated host and
monitor compliance sections; and the use InterfaceIndexOrZero instead of
redefining InterfaceIndex.

There are a number of open issues with the use of multiple MIB agents in
switches, the optional nature of the interface index and to permit easier
selection of session entries by session address which have to be resolved,
but it is expected that this will be complete within the next few weeks, at
which time a new draft will be issued ready for working group last call. A
reference implementation is available of the -02 draft, work is progressing
on the latest version. The RTP MIB will be referenced by H.341, the ITU's
H-series management specification.

The RTP payload format for DTMF and other telephone tones
(draft-ietf-avt-dtmf-01.txt, draft-ietf-avt-tones-00.txt) was presented by
Scott Petrack. The justification for using a new payload format is to
reproduce the tones better than low-rate codecs can do; to apply
redundancy differently for tones and speech; to separate detection of tones
>from their interpretation; and to preserve the sounds associated with
particular signals. Two payload formats are defined: a named signal event
payload and a tone frequency format payload. 

The named signal event payload includes a named event (dialtone, busy,
call-waiting, off-hook, etc), together with the volume and duration of that
event. The tone frequency format sends a representation of the actual set
of tones to be played, rather than their purpose (name), enabling a wider
range of tones and the playout of foreign tones. 

Both formats may be conveyed in a single packet, using RFC2198 redundancy,
for robustness. Mark Handley expressed some concern with the means by which
this is done: in particular it is unclear how to playout these tones when a
fraction of the tone is lost. 

The two drafts are to be merged within the next couple of months, and it is
the desire of the authors to move the result to the standards track soon
after. Some concern was expressed that this draft is not receiving
sufficient exposure in bodies such as the ITU which may be able to provide
relevant feedback - it must be ensured that these bodies have a chance to
comment before this document can progress in the IETF.

The discussion of RTP multiplexing started with a presentation by Jonathan
Rosenberg on issues in RTP multiplexing (draft-ietf-avt-muxissues-00.txt).
As noted in this draft, there are two scenarios which are of interest:
end-to-end and mid-network multiplexing. The scenario chosen will affect
the choices made in designing the multiplexing scheme. These choices affect
the delineation, identification, synchronisation and dynamism of the
multiplexed data.

There are three means by which data in a multiplexed packet may be
delineated: by explicit length indicators (which have maximum flexibility
but the largest overhead); implicitly based on payload type (many codecs
are fixed length blocks); or implicitly by out of band signalling (which
has the least bandwidth overhead, but requires signalling and makes
changing encodings or packetisation difficult).   If all multiplexed
packets are of the same duration this problem becomes much simpler.

Multiplexed data may be identified by an explicit ID, where the number of
bits used depends on the number of simultaneous calls and the desired ID
reuse latency. This ID may be the RTP SSRC in some cases. Alternatively the
data may be channelised, where each user's data gets a slot in the packet.
This latter approach requires out-of-band signalling and, of silence
suppression is used, a bitmask to indicate which channels are active. The
overheads of the two schemes vary with the number of channels being
multiplexed, whether channels are active continually, and with the rate of
change of the set of multiplexed streams.

If each frame of multiplexed data within a packet has the same timestamp,
the individual timestamps may be elided, and replaced by the single
timestamp in the multiplexed packet's RTP header. If the timestamps are
close, offsets relative to the outer timestamp may be used rather than
complete timestamps (saving some number of bits). Alternatively if users
have uncorrelated frame start times it is necessary to send the complete
timestamp per user.

The dynamism of codecs also affects the multiplexing format chosen. If
codecs never change there is no need to include a payload type indication
within the multiplexed stream, and out of band signalling may be used.
Similarly, as codecs change rarely, out of band signalling may still be
appropriate, depending on how often codecs may change (synchronisation
between the signalling of a payload type change and the media stream may be
complex). If codecs change frequently, some form of in-band payload type
indication is most appropriate - this need not necessarily be a complete
RTP PT value if the set of allowable codecs is small a mapping table may be
used instead.

The marker bit may need to be transmitted depending on the use of the
multiplexed stream.

It was noted that this is a very large space, and a number of solutions to
the multiplexing problem are possible. This group has a number of solutions
presented to it, yet the precise problem definition for each of these has
not been enunciated. It may help to focus the discussion if the question
"why are you multiplexing?" is clearly answered, and if we derive a number
of scenarios which require common solutions.

Essentially, we need to focus on requirements. Trying to do generic
optimisation using a multiplexer is futile.

The first multiplexing proposal (an update to draft-ietf-avt-mux-rtp-00.txt) 
was presented by Barani Subbiah. The stated goals of this proposal are to
achieve the best possible fit with cellular/PSTN applications and to derive
a payload format suitable for use in a switched IP telephone network. They
are aiming for a simple format with a fixed header suitable for hardware
implementation, providing a compromise between bandwidth saving (in
addition to the outer RTP header, this proposal averages two bytes overhead
per multiplexed stream) and complexity.

The sequence number (2 bits) is new since the last draft, this means a
reduction in the length field size to 5 bits. Concern was expressed that 
a 5 bit length field is insufficient for some audio codecs which may be
desirable. The use of the 2 bit sequence number was questioned, since 4
packet losses are possible -- a longer field should probably be used.

A transition bit is included to signal a change in the end-to-end flow
parameters, allowing one state change per RTT. It was noted that if all
packets with an RTT are lost, the state change will not be noted at the
receiver. It was also noted that payloads such as DTMF interspersed with
voice can cause change in payload type more often than once per RTT.

Tohru Hoshi presented draft-tanigawa-rtp-multiplex-01.txt. This is a simple
proposal where multiple RTP packets are concatenated for transmission in a
single UDP packet. The default packetisation interval specified for a codec
in the audio/video profile is used such that no length indication is
necessary (or can be signalled out-of-band, if a non-default interval is
desired). The draft now includes a section describing the efficiency gains
for using this proposal according to various metrics. Call setup signalling
is also defined.

The next multiplexing proposal discussed was GeRM (draft-ietf-avt-germ-00.txt),  
presented by Mark Handley. The goal of this proposal is to transparently
multiplex a number of RTP streams. It operates using difference coding
between the headers of packets to be multiplexed together. Clearly this
will work better if the packet headers are similar (this can be achieved
between cooperating gateways, although the traffic pattern will affect
performance) but it will still work if the end points do not cooperate,
and will perform no worse than simply concatenating packets.

The GeRM protocol is well suited for scenarios where a mix of RTP packets
are to be multiplexed, such as may be encountered in the transport of
MPEG-4 streams, or for use between a pair of cooperating gateways
multiplexing a large number of similar streams. It achieves considerable
flexibility, at the expense of complex parsing and greater bandwidth
overhead than other, less general, protocols. 

The final multiplexing presentation was produced by Dean Willis during the
meeting, so no draft exists.  The assumptions here are that large numbers
of streams are being carried between end-point pairs, fast interfaces with
minimal serialisation delay are used, and mixed codecs with silence
suppression exist. The goal is to increase overall "network efficiency" by
re-packing packets to increase the total MTU and reduce the number of
packets sent. Two alternatives were considered:
- RTP level: complex, no benefit to non-RTP traffic, issues with RTCP
- UDP level: brute force level simplicity, aggregate UDP flows between
  end-points, allows IPsec at multiplexing level, transparent to the
  application, allows mid-network multiplexing, less efficient
The UDP level approach is favoured.

Following the presentations a considerable amount of discussion ensued.
Concern was expressed that multiplexing is being used as an "RTP switching"
solution, with application level routing: it was noted that IP has a number
of perfectly reasonable routing algorithms already, and it is unnecessary
to re-invent these within RTP. Many people expressed concern that the
problem to be solved by multiplexing has not been clearly stated: is it to
reduce the number of packets sent? the number of bytes sent? to perform
application level RTP routing? etc. It is unclear that RTP multiplexing is
the correct solution here: a generic UDP multiplexing protocol (as in the
final proposal) may be more suitable in some cases.  Carsten Bormann
succinctly stated that if we are to define an RTP multiplexing scheme,
it should be an absolute requirement to preserve the integrity of the
RTP information.  If it does not, then it is not RTP multiplexing, it
is a new protocol.

Concern was expressed that multiplexing streams with different transport
level addressing into one is not clearly handled by these proposals. In
some cases, the SSRC is assumed to provide a unique stream ID, which is 
not necessarily the case across multiple streams. The handling of RTCP data
by a number of these proposals is also unclear.  The proposals need to
be extended to address these issues.

Some of the proposals specify a particular form of signalling in addition
to the payload format. These payload formats should be independent of the
signalling to be used. The proposals may want to express signalling
requirements, but should not tie the payload format to a specific scheme.

Since it is unclear whether a single protocol can satisfy multiple aims,
and which of the five proposals currently submitted to the group will
go forward, the authors and other working group participants are
requested to submit application scenarios in which multiplexing is to
be applied.  Within those scenarios, assumptions about traffic can be
made explicit.  We'll choose three or four scenarios and ask the
authors to simulation or analysis to quantify the performance of their
proposals under those scenarios, in order to facilitate a fair comparison.

The next subject was the transport of MPEG-4 streams within RTP. A number
of AVT members participated in the MPEG meeting in Atlantic City in
October, leading to the formation of an ad-hoc group within MPEG to discuss
MPEG transport using IP. The work conducted in that group to date was
presented by Reha Civanlar. 

A number of alternatives for the transport of MPEG-4 streams in an IP
network were considered: 
	- directly on UDP
	- RTP followed by a full MPEG-4 SyncLayer packet
	- MPEG-4 SyncLayer packets mapped onto RTP packets
	- MPEG-4 elementary streams over RTP with natural payload formats
The preferred approach would be to use the latter approach, but since the
ES interface is not a normative part of MPEG-4 this may not be feasible.

The approach chosen is, therefore, to map the MPEG-4 SyncLayer packets onto
RTP packets, such that the common pieces of the header reside in the RTP
header, with a small payload header providing the MPEG-4 specific features.
A single payload format is used for MPEG-4 streams transported within RTP,
and the MPEG-4 model is maintained (although not the precise packet
format). In this approach, an RTP multiplexing scheme is needed to fulfill
the role of FlexMux in MPEG-4. The GeRM proposal seems to be a good fit for
this.

An internet draft detailing this work is in preparation. Those who wish to 
participate in this work are encouraged to join the ad-hoc group's mailing
list: send email to 4onIP-sys-request@fzi.de in order to subscribe.

Christine Guillemot presented an RTP generic payload with scalable and
flexible error recovery (draft-guillemot-genrtp-00.txt). This draft takes
a somewhat different view of the problem of transporting MPEG-4 content and
is based on carrying elementary streams in a generic manner. The motivation
for this is to transport many types of stream whilst avoiding having to
define a payload format for each and allowing finer control of error
correction with a set of different FEC mechanisms and the possibility of
grouping AUs in a single packet.

One of the aims of this work are to factorize the common features instead
of developing specific formats for each codec/type of elementary stream and
to be able to identify repeated data so that the network adaptive layer can
identify and remove this if desired. The adaptation layer can add FEC to
entire packets or to portions of a stream within packets (adding redundancy
in a similar manner to RFC2198).

Concern was expressed that adding FEC to portions of a packet adds a lot of
extra complexity, and unless this FEC is much smaller than that which would
otherwise be present this complexity may not be justified.

Some concern was expressed that this document includes a number of payload
formats (redundancy, FEC, fragmentation and grouping) which may be better
separated. This clearly depends on the details of the stream which is being
packetised.

It is unclear that this format is suitable as a generic RTP payload
independent of MPEG-4, however it may work well as a general purpose
transport for MPEG-4 elementary streams.

Steve Casner described the changes made to the main RTP specification since
the last meeting. This is now stable, and unless major problems are found
is believed ready for last-call for draft standard. The changes since the
last meeting are described in draft-ietf-avt-rtp-new-02.txt and include:
	- SSRC sampling moved to separate draft (ietf-avt-rtpsample-01.txt)
	- Keep only unconditional reconsideration 
	- Add IANA considerations section added; no longer suggest experimental
  	  registration of values
	- Y2036 (in)consequences explained
	- convert to MUST, SHOULD, MAY
A plea was made for help checking:
	- Section 0: resolution of open issues
	- Section 6.2: RTCP transmission interval
	- Section 6.3: RTCP send and receive rules
	- Appendix A: does the code work?
	- Appendix B: changes from rfc1889
It was noted that the group must document "at least two independent and
inter-operable implementations from different code bases" of "all of the
options and features of the specification" in order to advance to draft
standard status (RFC2026). Colin Perkins volunteered to produce a draft
detailing those options and features as a checklist for vendors to check
compliance, and Jonathan Rosenberg volunteered to produce a draft detailing
tests for the timer reconsideration algorithms.  Since the meeting,
Jonathan has done a careful check of the code in Appendix A and found
several problems to be fixed.

The changes to the RTP profile (draft-ietf-avt-profile-new-04.txt) since
the last meeting are less advanced: a clearer statement of the new policy
of no more static assignments, and the addition of change bars. It is
still necessary to complete the update with MUST, SHOULD, MAY, etc and to
add text to allow default of 5% RTP bandwidth to be overridden. 

The registration of RTP payload format names as MIME types is still not
complete: Philipp Hoschka volunteered to work on this, and to work out the
details of the process. It is hoped that this may just be a statement we
can put in the profile to specify how the registration is done, without
changing the MIME registration process, but this is not yet clear.  

The working group has agreed in previous IETF meetings that any
additional RTCP SDES items should be defined in separate RFCs rather than
adding them to the base RTP spec.  This is in part to minimize changes
at the transition from Proposed to Draft Standard but also because we
did not want implementors to infer that all applications should
include all the SDES items.

Accordingly, Peter Parnes has written draft-parnes-rtp-sdes-00.txt to
propose the addition of new SDES items Nickname, Homepage,
Personal_image and Active_media.  Steve Casner presented this proposal
since the author could not attend.  This proposal is similar to the
set of potential new SDES items discussed at the 41st IETF in Los
Angeles, though Organisation was included previously and Active_media
is new here.  Comments are requested from the group as to whether this
is the right set of new SDES items to define at this point, or whether
some others should be added or some of these deleted.  The status of
this RFC would be Experimental rather than Proposed Standard.

Steve Casner repeated a concern expressed previously that about the
inclusion of URLs in an SDES packet which may be sent to a large
multicast group, since simultaneous retrieval of these by many
receivers can cause implosion problems. This draft specifies that
retrieval should either be done only in response to direct user action
or if automated should be delayed by a random interval (after receipt
of the RTCP packet).  Is this specification sufficient?

Mark Handley asserted that we shouldn't try to turn RTCP into a
general data transfer mechanism, but did favor adding Organization
since current practice is to include that information in the Name
item.  The consensus of the group was that it was reasonable to make
extensions to RTCP, but the additional information must be optional,
and should not be required for operation of the application.
Extension RFCs should include an applicability statement for each
item.  Further comments on this draft will be requested on the mailing
list to establish consensus for proceeding.

The SSRC sampling algorithms (draft-ietf-avt-rtpsample-01.txt) were
presented by Jonathan Rosenberg. These have been moved out of the main 
RTP specification because of the IPR issues (Lucent patent on the binning
algorithm). The changes to this document are:
	- Uniformity of SSRC values usage: recommend hashing SSRC value,
	  because some broken implementations doesn't choose uniformly
	  distributed SSRC values
	- New section on performance of sampling in terms of coefficient
	  variation added
	- An explicit statement of the IPR issues and licensing terms needs
	  to be added
Comments are requested. 

The document giving guidelines for writers of RTP payload formats
(draft-ietf-avt-rtp-format-guidelines-01.txt) was noted as being
essentially complete. One more revision will be produced to include
some comments recently received, so others wishing to make comments
should do so as soon as possible.  That revision will be last called
for BCP status.

The payload format for PureVoice(TM) audio (draft-mckay-qcelp-01.txt) was
noted as being in working group last call still, pending resolution of a
question regarding the proposed scheme for encryption of the payload
data in a non-standard manner.  A compromise solution has been worked
out in offline discussions with the authors, so a revision of the
draft is expected soon so last call can be completed if there are no
further objections.

The Generic FEC payload (draft-ietf-avt-fec-04.txt) was presented by
Jonathan Rosenberg. Changes include the removal of Reed-Solomon coding 
(to a separate draft) and mask extension. The examples and code have
been tested and bug-fixed, and the issues with encryption during key
changes have been resolved. This document is essentially ready for last
call as proposed standard - will do one more revision to get the MUST,
SHOULDs, etc sorted. Comments are sought.

The Reed-Solomon draft (draft-ietf-avt-reedsolomon-00.txt) was also
presented by Jonathan Rosenberg. Help is required with this: if anyone
having expertise on the Reed-Solomon algorithm is interested in seeing this
work progress they should contact Jonathan, else the draft will not be
updated.

A proposal to use the RFC2198 redundancy format as a transport for
interleaved audio (draft-ietf-avt-interleaving-00.txt) was presented by
Colin Perkins. This may eventually be merged into that document, although
this is at present undecided. Comments are sought.

The meeting concluded with a brief presentation on the proposed charter
revision. It was noted that this needs clarification regarding MPEG-4
payload formats, but is otherwise satisfactory. The revised charter
will be sent for IESG approval in the near future.

-------------------------------------------------------------------------------



From rem-conf Thu Jan 07 19:12:27 1999 
From rem-conf-request@es.net Thu Jan 07 19:12:25 1999
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From: "Michael W. Fox" <mfox@nuera.com>
To: <petrack@vocaltec.com>
Cc: <rem-conf@es.net>
Subject: Re: draft-ietf-avt-tones-00.txt
Date: Thu, 7 Jan 1999 19:04:44 -0800
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	boundary="----=_NextPart_000_01A2_01BE3A70.961661F0"
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Regarding the named events, specifically those used for call progress =
indication, I think that it would be useful to turn a named indication =
on for a long duration then turn it off sometime less than the duration =
later. This is useful for something like ringback, where one can not =
predict when the answer will occur. I apologize if this is in the draft, =
or discussed in Orlando, and I missed it.

Best Regards,
Mike Fox

------=_NextPart_000_01A2_01BE3A70.961661F0
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD W3 HTML//EN">
<HTML>
<HEAD>

<META content=3Dtext/html;charset=3Diso-8859-1 =
http-equiv=3DContent-Type>
<META content=3D'"MSHTML 4.72.3110.7"' name=3DGENERATOR>
</HEAD>
<BODY bgColor=3D#c8e0d8>
<DIV><FONT size=3D2>Regarding the named events, specifically those used =
for call=20
progress indication, I think that it would be useful to turn a named =
indication=20
on for a long duration then turn it off sometime less than the duration =
later.=20
This is useful for something like ringback, where one can not predict =
when the=20
answer will occur. I apologize if this is in the draft, or discussed in =
Orlando,=20
and I missed it.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#000000 size=3D2>Best Regards,</FONT></DIV>
<DIV><FONT size=3D2>Mike Fox</FONT></DIV></BODY></HTML>

------=_NextPart_000_01A2_01BE3A70.961661F0--




From rem-conf Fri Jan 08 09:17:52 1999 
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Date: Fri, 08 Jan 1999 09:09:45 -0800
To: mbone@ISI.EDU, rem-conf@es.net
From: Marty Bickford <martinb@stardust.com>
Subject: White Paper: Managing IP Multicast Traffic
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Hey Everyone,

Kevin Almeroth has completed a white paper called "Managing IP Multicast
Traffic". 

The paper is supplied by IPMI as a part of the 3rd Annual IP Multicast
Summit on Feb 7-9, 1999 in San Jose, CA USA.

It is available at:
 
 http://www.ipmulticast.com/events/summit99/  or 
 http://www.ipmulticast.com/events/summit99/whitepaper.htm
 
If you haven't registered with IPMI, you'll be asked to.

Sincerely,
Marty
____________________________________________________
Marty Bickford  - 408.879.8080 (8081-fax)
Stardust Forums - http://www.stardust.com

3rd Annual IP Multicast Summit - Feb 7-9, 1999
http://www.ipmulticast.com/events/summit99/



From rem-conf Fri Jan 08 10:09:53 1999 
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Date: Fri, 8 Jan 1999 11:30:00 -0400
To: Tempting Tear-Outs <temptingtearouts@1stconnect.com>
From: Tempting Tear-Outs <temptingtearouts@1stconnect.com>
Subject: >>>>>>  L@@K!    Amazing Free Offer!!!   Unbelievable, but True!
 <<<<<<<    82
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To be removed from our mailing list, please send email to:
temptingtearouts@1stconnect.com with the subject line of "remove."

FOR MORE INFO:   please "cut out" the below form on the "cut" lines shown,
and fax it, for the fastest reply to:                1-718-227-9125   (this
is a fax # in the USA)

or send via smail (first class mail or airmail) to:
                                         Tempting Tear-Outs
                                         Att. Free-catalogue-by-email Dept
                                         3835 Richmond Ave.  Suite #200
                                         Staten Island NY  10312-3828
                                         USA

SORRY, BUT.... our software is not set up to accept the below form via
return email;   WE CAN ONLY acknowledge forms sent in via fax or smail.

--> IMPORTANT complete directions, to ensure that you get a reply, and more
info follow, below the reply form and the catalogue options.


*------------cut here/begin-------------------------------------------*

Name (First Middle Last):
Internet email address:
Smail home address:
City-State-Zip:
Country:
Work Tel. #:
Work Fax #:
Home Tel. #:
Home Fax #:
Cellular (Mobile) Tel. #:
Beeper (Pager) Tel. #:

How did you hear about us (name of person/company who referred you or the
area of
the internet that you saw us mentioned in):  Referred by:  Tempting
Tear-Outs
010699-ls82-lafoubt-la

Name of USA mags you currently get on the newsstand or in the store:

Name of USA mags you currently get on a subscription basis, through the mail:

Name of USA mags you would like price quotes on when we call you:

Catalogue version desired (list number of choice below):

*------------cut here/end--------------------------------------------*



CATALOGUE VERSION CHOICES:

1.  This version can be read by everyone, no matter what type of
     computer you use, or what type of software you use.  It is a simple
     format, with just our entire catalogue pasted into the body of a
     single email message, 316K in size.  If you use pine or elm on a unix
     system or an advanced software version such as Eudora Pro 3.0 or
     later, you will most likely receive it as a single email message.
     However, if your software limits incoming email messages to a
     certain size, say 32K or so, then your software will split it into
     multiple email message parts.   Whether you receive it as a single
     email message or multiple part email messages, you can easily
     paste it into one whole text document with your word processor, in
     about 10 minutes or so.
2.  For more advanced computer users:  attached plain ascii text file
     ~316K - you must know how to download an attached text file and
     then be able to locate it on your hard drive or system home
     directory;  it can then be opened with any pc or mac word processing
     software.  If in doubt, don't ask for this version.  This isn't for
     internet *newbies.* Better to order option 1 and spend a few minutes
     pasting them into one whole text document with your word processor,
     than to waste hours trying to figure how to deal with this option.
     This version is great for doing keyword searches and jumping around
     within the catalogue with your word processing software, if your
     normal email reading software doesn't allow this.


VERY IMPORTANT DIRECTIONS TO ENSURE THAT YOU GET A REPLY:

1.   you must call from an "unblocked number," ie. one that is not blocked
>from caller id.  We are very sorry for this requirement, but our fax
software requires this before it allows an incoming fax call to connect.
If you have a blocked number, you must first unblock it.  In most cases
this means dialing *82 from a touch-tone phone (or 1182 from a rotary
phone) before you dial 1-718-227-9125.     NOTE:  If you are not sure if
your number is blocked, just try dialing our fax # normally.  If you don't
get a recording telling you your number is blocked, your number has been
transmitted and you may press the start button on your fax when you hear
the fax tone from our fax.
2.   no reply forms can be accepted by email....only via fax or smail.
3.   your form must be typewritten or printed out on your computer printer
before you fax it;  sorry, but *no* handwritten forms will be acknowledged.
If you can't find someone with a typewriter or a computer printer, we
apologize for not being able to reply to you.
4.   faxes with cover pages will be rejected.  You must send *only* the
reply form.
5.   forms not *completely* filled in will not be acknowledged.
6.   you will receive a reply within 1 business day directly from the
company making the offer via email.  Therefore you must have an email
address.  If you read this message, then you must have an email address, or
access to one, at least.   :-)
7.   your fax must not exceed 1 page in length.   Faxes of 2 or more pages
will be sensed, then auto-terminated and deleted.  Your fax goes directly
onto our 5.0 gigabyte hard drive and we must limit all incoming faxes to 1
page.
8.   all faxes must begin with:
*------------cut here/begin-------------------------------------------*
and must end with:
*------------cut here/end--------------------------------------------*
9. Any fax not conforming to this format will be sensed by our software,
then auto-terminated and deleted from the hard drive, before any human ever
gets to see it.
10. The type on your fax must be dark and legible.   If in doubt, please
print it out darker before faxing it in.  If we can't read it, we can't
reply to you or send you our FREE catalogue.     :-(
11.  If this all seems too complicated for faxing, just do it the old
fashioned way via smail!!!


WHO WE ARE:

Tempting Tear-Outs is an advertising company that brings potential new
customers to the companies they advertise for.

MORE ABOUT THE COMPANY MAKING THE FREE OFFER:

The company making the offer is a magazine subscription agency based in the
USA.  They have over 1,100 popular USA titles available to be shipped to
*any* country, including of course, to anywhere in the USA!    They offer a
FREE 1 yr. subscription to your choice of over 200 of the titles in their
catalogue to any new customer using them for the first time.       The
dollar value of the freebies, based on the subscription prices directly
>from the publishers, ranges from $6.97 all the way up to $50.00!

For new customers in the USA, there is no charge for FPH (foreign postage &
handling), so the freebie is 100% free!   For new customers living
overseas, the only charge on the freebie would be for the FPH (foreign
postage & handling).

Their president has been in the magazine subscription business since 1973
and they are very customer-service oriented.   They will even help you with
address changes on your magazines, even if you move from one country to
another country.   They have thousands of happy customers in over 59
countries.

Their price guarantee is very simple:       they guarantee that their
subscription prices are the lowest available and they will BEAT any
legitimate, verifiable offer before you pay them or match it afterwards, by
refunding you the difference in price PLUS the cost of the postage stamp
you would use sending in the special offer to them, even 6 months after you
pay them, as long as it was current at the time of your offer.    Does that
sound fair?       Wouldn't it be great if everything you bought came with
that price guarantee?

Sometimes they are less than half of the next best deal out there,
sometimes just a little cheaper, but always you get the lowest rates
without having to shop around.     With 1,100+ titles on their list, they
would like to think that they have also the best selection around!

Within the USA, for their USA customers, they are cheaper than all their
competitors and even the publishers themselves.  This is their price
guarantee.         The 1 yr. freebie that you get with your first order is
completely free!

Overseas, (even after you factor in the cost of the FPH (foreign postage &
handling) and the conversion from USA Dollars to your currency), on the
average, they are generally around one-fourth to one-half of what the
newsstands overseas charge locally for USA magazines.  On some titles they
are as little as one-tenth of what the newsstands charge.  They are also
the cheapest subscription source for delivery overseas, including directly
>from the publishers themselves!   Some publishers don't even offer
subscriptions overseas.........but overseas subscriptions are this
company's specialty!  They feel that magazines should not be a luxury
overseas.   In the USA, people buy magazines and then toss them after
reading them for just a few minutes or hours.  They are so cheap in the
USA!   Well, this company would like to make it the same way for their
overseas customers.  They are also cheaper than all their competitors in
the USA and overseas, including the publishers themselves!   It is also
*highly unlikely* you will find any of their USA competitors calling you
overseas, in order to offer that personal touch, just to sell you a couple
of magazines!  But that is what this company specializes in and loves
doing!     Around one-half their business comes from overseas, so they are
very patient with new customers who only speak limited English as a 2nd
language.    Subscription prices quoted for overseas consist of the
subscription price, plus the FPH.   You add the two together and that is
your total cost.   The exception is the 1 yr. freebie you get with your
first order.   On that title, you pay *only* the FPH for the 1 yr. term.

Their prices are so cheap because when you deal with them, you cut-out all
the middlemen.


HERE IS HOW YOU CAN GET MORE INFO AND GET STARTED WITH THEM:

Simply fax or smail back to us the reply form listed at the top of this
message.   We will then forward your form on to the subscription agency.
They will then email their "big and juicy" catalogue to you, in whichever
of the two formats you chose.   The catalogue is FREE and makes for hours
of fascinating reading, on its own. It includes the complete list of
freebies, a complete list of all the titles they sell, as well as detailed
descriptions on most of the titles, along with lists of titles by category
of interest and their terms of sale.

They will then give you a friendly, no-pressure, no obligation, 5-minute
call to go over how they work and to answer any questions that you might
have, as well as give you up-to-the minute price quotes on any titles you
might be considering.     They will call you in whatever country you live
in, taking the time difference into account.        As they like to
emphasize the personal touch they give to each new customer, all first-time
orders can only be done via phone, so they can answer all your questions
completely and personally.   Once you have placed your first order via
phone, you will be able to place future orders and make inquiries on your
account, get price quotes, etc., all via email, if that is most convenient
for you.

Within the USA, they accept payment via check over the phone, Mastercard,
Visa, American Express, Diner's Club and Carte Blanche.    Overseas, they
accept Mastercard, Visa, American Express, Diner's Club and Carte Blanche,
even if your credit card is a local one in local currency (that most
merchants in the USA would not normally be willing to accept).

That's our introduction of our client that we represent.   We hope that we
have piqued your interest and that you will take the next step to get their
free catalogue!   Thank you for your time and interest.

--
Tempting Tear-Outs.
For more info on advertising rates, please write us on your company
letterhead, w/business card, via smail to:   Tempting Tear-Outs, 3835
Richmond Ave. Suite #200, Staten Island NY  10312-3828, USA.





From rem-conf Fri Jan 08 10:54:23 1999 
From rem-conf-request@es.net Fri Jan 08 10:54:21 1999
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Date: Sat, 9 Jan 1999 04:49:38 +0100
From: Compte idms99 <idms99@ensica.fr>
Message-Id: <199901090349.EAA06342@magic>
To: rem-conf@es.net
Subject: IDMS99 call for papers
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			C a l l    f o r    p a p e r

				  IDMS'99
		  6th International Workshop on Interactive
		     Distributed Multimedia Systems and 
		         Telecommunication Services

		    12-15 October 1999, Toulouse, France


                                Organized by
                            LAAS-CNRS and ENSICA
                         http://www.ensica.fr/idms99
                          email : idms99@ensica.fr

The Sixth International Workshop on Interactive Distributed Multimedia 
Systems and Telecommunication Services will be held in 1999 at ENSICA, 
Toulouse, France. The purpose of this workshop is to bring together 
researchers, developers, and practitioners from academia and industry. 
The workshop serves as a forum for discussion, presentation, and 
exploration of technologies and their advances in the broad field of 
interactive distributed multimedia systems and telecommunication 
services - ranging from basic system technologies such as networking 
and operating system support to all kinds of teleservices and distributed 
multimedia applications. Case studies and papers describing experimental 
work are especially welcome. Relevant topics include, but are not 
limited to: 

* High-speed/ATM networks 
* Mobile multimedia systems 
* Multimedia over satellite 
* Multimedia middleware 
* Quality of service issues 
* Media scaling 
* Resource management 
* Protocol design and implementation 
* Development tools for distributed multimedia applications 
* Distributed multimedia database systems
* Multimedia-specific intelligent agents 
* Computer supported collaborative work 
* Distributed virtual reality systems 
* Distance education 
* Conferencing 
* Digital libraries 
* Interactive television 
* Video-on-demand systems 
* Compression algorithms 


IDMS'99 will consist of a three day technical program, a full day of 
tutorials, and demonstrations during the workshop. In order to keep the 
flavor of a workshop, the number of participants will be restricted. 
Furthermore, we encourage contributions in form of full papers and 
position papers. Full papers are expected to describe innovative and 
significant work. The purpose of position papers is to provide a seed 
for debate and discussion. Position papers enable researchers to 
present exciting ongoing work in early stages, suggestions for 
future directions, and concerns about current developments. Both types 
of papers will be reviewed by the program committee and printed in the 
workshop proceedings edited by Springer Verlag in the Lecture Notes in 
Computer Science series.


Information for authors: Authors are invited to submit full papers and 
position papers for review. Submitted manuscripts must describe original 
work (not submitted or published elsewhere). Full papers must not be 
longer than 20 double spaced pages and position papers must not be longer 
than 8 double spaced pages. Both types of papers should contain an abstract 
of approximately 300 words, and include title, authors and affiliations. 
The submission process of papers will be handled electronically. Detailed 
description of the electronic submission procedures is available on the 
IDMS'99 web page: http://www.ensica.fr/idms99/. Authors without web 
access may email to owe@laas.fr for requesting electronic submission 
information. Authors unable to submit electronically are invited to 
send 5 copies of their contribution to Dr Philippe Owezarski.

Important Dates
Submission due:			March 8, 1999
Notification acceptance:	April 30, 1999
Camera ready version:		June 14, 1999
Workshop:			October 12 - 15, 1999


Co-chairs: Michel Diaz, Philippe Owezarski, Patrick Sיnac

LAAS-CNRS, 7 Avenue du Colonel Roche, 31077 Toulouse cedex 4, France
e-mail: {diaz, owe}@laas.fr, Phone: +33 5 61 33 63 17, Fax: +33 5 61 33 64 11

ENSICA, 1 Place Emile Blouin, 31056 Toulouse cedex, France
e-mail: senac@ensica.fr, Phone: +33 5 61 61 86 81, Fax: +33 5 61 61 86 86



Program Committee Chairmen:		Michel Diaz, Philippe Owezarski
Organization Chairman:			Patrick Sיnac
Tutorial Chairman: 			Laurent Dairaine
Publicity and Demonstration Chairman: 	Pierre de Saqui-Sannes
Conference electronic support Chairman:	Marc Boyer

Program Committee

H. Affifi, ENST Bretagne, France
P.D. Amer, U. Delaware, USA
E. Biersack, Institut Eurיcom, France
G. v. Bochmann, U. Ottawa, Canada
B. Butscher, GMD FOKUS, Germany
A.T. Campbell, Columbia U., USA
T.S. Chua, U. Singapore, SG
J.-P. Courtiat, LAAS-CNRS, France
J. Crowcroft, UCL, UK
L. Delgrossi, U. Piacenza, Italy
C. Diot, Sprint ATL, USA
R. Dssouli, U. Montrיal, Canada
M. Dudet, CNET France Tיlיcom, France
W. Effelsberg, U. Mannheim, Germany
F. Eliassen, U. Tromsר, Norway
S. Fdida, LIP6, France
D. Ferrari, U. Piacenza, Italy
V. Goebel, UNIK, Norway
T. Helbig, Philips, Germany
J.-P. Hubaux, EPFL, Switzerland
D. Hutchison, Lancaster U., UK
W. Kalfa, TU Chemnitz, Germany
T.D.C. Little, Boston U., USA
E. Moeller, GMD FOKUS, Germany
K. Nahrstedt, U. Illinois, USA
G. Neufeld, UBC Vancouver, Canada
B. Pehrson, KTH Stockholm, Sweden
T. Plagemann, UNIK, Norway
B. Plattner, ETH Zurich, Switzerland
L.A. Rowe, U. California Berkeley, USA
H. Scholten, U. Twente, Netherlands
A. Seneviratne, UTS, Australia
R. Steinmetz, GMD, Germany
J.B. Stיfani, CNET France Tיlיcom, France
L. Wolf, TU Darmstadt, Germany
M. Zitterbart, TU Braunschweig, Germany



From rem-conf Fri Jan 08 12:36:38 1999 
From rem-conf-request@es.net Fri Jan 08 12:36:37 1999
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	Fri, 8 Jan 1999 15:53:44 -0500 (EST)
Date: Fri, 8 Jan 1999 15:53:44 -0500 (EST)
From: <ladylezah@hotmail.com>
To: <poplar@oanet.com>
Message-Id: <18161.236168.63074167 sales@gky.com.au>
Subject:  FROM THE DESKTOP OF MAY.....
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I tried them all and lost money...Does that sound familiar?
Well, guess what? I finally found a program that works!!!

WAIT A MINUTE! Before you say, " here's another one " 
and want to delete this: I urge you to read this through!
This is changing my life for the better and it certainly
will do the same for you!

                                                ***** Visit our website at: http://www.topsecrets100.com *****
                                                                                        CODE:59441
______________________

Pros Make You a Fortune.

The # 1 Homebased Business for the 2nd year in a row.
No selling, just refer people to the company'S 800 #.
Commission checks paid weekly! It's simple and easy.
$500 to $5000 (or more) monthly! A solid company that's 
a member of the Better Business Bureau and the United Chamber of
Commerce.

HAVE YOU EXPERIENCED THIS?

The amount of money earned in your program is only a
fraction related to the MASSIVE AMOUNTS OF TIME,
ENERGY AND MONEY YOU PUT OUT! We all have,
right? Why?

" PEOPLE DO NOT LIKE TO SELL, NEVER HAVE
AND NEVER WILL "

MOST PEOPLE ARE NOT INTERESTED IN:

#1. SELLING ANYTHING TO ANYONE!
#2. FOLLOWING UP BY E-MAIL, SNAILMAIL, OR PHONE!
#3. MONTHLY REQUIRED PURCHASES!
#4. RECRUITING ON AND OFF THE NET!
#5. MAKING PERSONAL CONTACT WITH OTHERS!
#6. CONFERENCE CALLING!
#7. MEETINGS!
#8. FILLING OUT APPLICATIONS!
#9. KEEPING AN INVENTORY!

SO NOW WHAT? HOW CAN WE EVER BE SUCCESSFUL?

HERES HOW:REFER people to this TOLL - FREE NUMBER
(where Professional Telemarketers will PRESENT, INFORM,
ENTERTAIN, AND DO whatever is necessary to SIGN UP
YOUR PROSPECTS and place them in your downline for
you!) That's all you do period! If you can do that (REFER
PEOPLE TO THE 800 NUMBER, ON and or OFF the NET)
you could be successful beyond your wildest dreams!

YOU OWE IT TO YOURSELF TO GET THE WHOLE STORY:

CALL: 8am - 10pm  CST Monday - Saturday
1-800-811-2141 Code: 59441 or call 1-800-226-0633 Code: 59441
Fax On Demand: 415-273-6020 (7 pgs.)
Outside the USA and Canada call:
1-785-762-6715 Code: 59441 or 1-785-762-2442   Code: 59441

HOW WELL DOES IT WORK?

Here are just a few Toll - Free recorded testimonials from company
members. Don't just take my word for it...call them for yourself! They
are real people just like you and me!

CALL NOW: THE CALLS ARE FREE!

Steve Fletcher: 888-438-4005
Jill Dietch: 888-703-5389
Jim Montgomery: 888-446-6949
William Sounder: 888--446-6951
Henry Wheeler: 888-256-4767
Tim Russell: 888-731-3457

Live operators will be happy to explain the program to you
and why it virtually GUARANTEES your success!

EVERY ASPECT OF SELLING IS DONE COMPLETELY BY
THE COMPANY and YOU GET PAID FOR IT! The company
handles all your calls, closes your sales and sends you
WEEKLY commission checks!

You receive $100 EVERY TIME THE COMPANY CLOSES A 
SALE FOR YOU!

PLUS you become eligible to receive the GUARANTEED
LOWEST PRICE on over 250,000 Name Brand Products and
Services you're already buying now, conveniently delivered
to your front door! SAVE $200 on groceries, $300 on dining
out and save hundreds more on THOUSANDS of name
brand products and services! Merchandise, Travel, Health
Care, Professional Services, Discounts on food and 
Entertainment and much more! The manufacture's
Warrantee is automatically doubled when you make a
purchase. DOUBLE THE DIFFERENCE MONEY - BACK
GUARANTEE you will save money using our program or
we will honor a FULL REFUND! Purchasing from among
these 250,000 Name Brand super discounted products and
services IS NOT MANDATORY.

YOU DON'T HAVE TO BUY ANYTHING.

You may contact people and refer them to the Toll - Free
800# any way you choose: Classifieds, flyers, postcards
(supplied by the company - including pre - qualified names),
Posting in appropriate newsgroups, Bulk e-mail etc.
Suppose you use Bulk e-mail, which I find very successful:
100,000 e-mails with a 1/10 of a percent (0.1%) sales rate,
nets, $1000...not to bad. Also, Top Secrets pays up to 
60% in bonuses to infinity! The most lucrative infinity
plan ever made.

When you call, consider the professionalism of the live
operators handling your call. I'm convinced you will not 
find a more highly motivated and well trained team of
people to close your sales for you, and remember they
are available 14 hours a day - 6 days a week
ON YOUR BEHALF!

CALL: 8am - 10 pm  CST Monday - Saturday
1-800-811-2141 Code:59441 or call 1-800-226-0633 Code: 59441
Fax On Demand:  415-273-6020 (7 pgs.)
Outside the USA and Canada call:
1-785-762-6715 Code:59441 or 1-785-762-2442  Code: 59441

Get your own CODE# from the friendly operators and I'll
RACE YOU TO THE BANK! DON'T DELAY! MAKE THAT CALL!
I promise you'll be glad you did!
                         Hazel

********* Visit our website at: http://www.topsecrets100.com *********
                                           Code: 59441

P.S. After you come on board give me a call at: 317-722-7707




From rem-conf Fri Jan 08 15:06:46 1999 
From rem-conf-request@es.net Fri Jan 08 15:06:44 1999
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	id 0zykvS-0005qM-00; Fri, 8 Jan 1999 15:02:42 -0800
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	Fri, 8 Jan 1999 18:24:52 -0500 (EST)
Date: Fri, 8 Jan 1999 18:24:52 -0500 (EST)
From: <ladylezah@hotmail.com>
To: <phlfreedom@aol.com>
Message-Id: <18161.236168.73565382 sharon-isaiah43@worldnet.att.net>
Subject:  FROM THE DESKTOP OF AL......
Mime-Version: 1.0
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Content-Transfer-Encoding: 7bit
X-Mailing-List: <rem-conf@es.net> 
X-Loop: rem-conf@es.net
Precedence: list

I tried them all and lost money...Does that sound familiar?
Well, guess what? I finally found a program that works!!!

WAIT A MINUTE! Before you say, " here's another one " 
and want to delete this: I urge you to read this through!
This is changing my life for the better and it certainly
will do the same for you!

                                                ***** Visit our website at: http://www.topsecrets100.com *****
                                                                                        CODE:46948
______________________

Pros Make You a Fortune.

The # 1 Homebased Business for the 2nd year in a row.
No selling, just refer people to the company'S 800 #.
Commission checks paid weekly! It's simple and easy.
$500 to $5000 (or more) monthly! A solid company that's 
a member of the Better Business Bureau and the United Chamber of
Commerce.

HAVE YOU EXPERIENCED THIS?

The amount of money earned in your program is only a
fraction related to the MASSIVE AMOUNTS OF TIME,
ENERGY AND MONEY YOU PUT OUT! We all have,
right? Why?

" PEOPLE DO NOT LIKE TO SELL, NEVER HAVE
AND NEVER WILL "

MOST PEOPLE ARE NOT INTERESTED IN:

#1. SELLING ANYTHING TO ANYONE!
#2. FOLLOWING UP BY E-MAIL, SNAILMAIL, OR PHONE!
#3. MONTHLY REQUIRED PURCHASES!
#4. RECRUITING ON AND OFF THE NET!
#5. MAKING PERSONAL CONTACT WITH OTHERS!
#6. CONFERENCE CALLING!
#7. MEETINGS!
#8. FILLING OUT APPLICATIONS!
#9. KEEPING AN INVENTORY!

SO NOW WHAT? HOW CAN WE EVER BE SUCCESSFUL?

HERES HOW:REFER people to this TOLL - FREE NUMBER
(where Professional Telemarketers will PRESENT, INFORM,
ENTERTAIN, AND DO whatever is necessary to SIGN UP
YOUR PROSPECTS and place them in your downline for
you!) That's all you do period! If you can do that (REFER
PEOPLE TO THE 800 NUMBER, ON and or OFF the NET)
you could be successful beyond your wildest dreams!

YOU OWE IT TO YOURSELF TO GET THE WHOLE STORY:

CALL: 8am - 10pm  CST Monday - Saturday
1-800-811-2141 Code: 46948 or call 1-800-226-0633 Code: 46948
Fax On Demand: 415-273-6020 (7 pgs.)
Outside the USA and Canada call:
1-785-762-6715 Code: 46948or 1-785-762-2442   Code: 46948

HOW WELL DOES IT WORK?

Here are just a few Toll - Free recorded testimonials from company
members. Don't just take my word for it...call them for yourself. They
are real people just like you and me!

CALL NOW: THE CALLS ARE FREE!

Steve Fletcher: 888-438-4005
Jill Dietch: 888-703-5389
Jim Montgomery: 888-446-6949
William Sounder: 888--446-6951
Henry Wheeler: 888-256-4767
Tim Russell: 888-731-3457

Live operators will be happy to explain the program to you
and why it virtually GUARANTEES your success!

EVERY ASPECT OF SELLING IS DONE COMPLETELY BY
THE COMPANY and YOU GET PAID FOR IT! The company
handles all your calls, closes your sales and sends you
WEEKLY commission checks!

You receive $100 EVERY TIME THE COMPANY CLOSES A 
SALE FOR YOU!

PLUS you become eligible to receive the GUARANTEED
LOWEST PRICE on over 250,000 Name Brand Products and
Services you're already buying now, conveniently delivered
to your front door! SAVE $200 on groceries, $300 on dining
out and save hundreds more on THOUSANDS of name
brand products and services! Merchandise, Travel, Health
Care, Professional Services, Discounts on food and 
Entertainment and much more! The manufacture's
Warrantee is automatically doubled when you make a
purchase. DOUBLE THE DIFFERENCE MONEY - BACK
GUARANTEE you will save money using our program or
we will honor a FULL REFUND! Purchasing from among
these 250,000 Name Brand super discounted products and
services IS NOT MANDATORY.

YOU DON'T HAVE TO BUY ANYTHING.

You may contact people and refer them to the Toll - Free
800# any way you choose: Classifieds, flyers, postcards
(supplied by the company - including pre - qualified names),
Posting in appropriate newsgroups, Bulk e-mail etc.
Suppose you use Bulk e-mail, which I find very successful:
100,000 e-mails with a 1/10 of a percent (0.1%) sales rate,
nets, $1000...not to bad. Also, Top Secrets pays up to 
60% in bonuses to infinity! The most lucrative infinity
plan ever made.

When you call, consider the professionalism of the live
operators handling your call. I'm convinced you will not 
find a more highly motivated and well trained team of
people to close your sales for you, and remember they
are available 14 hours a day - 6 days a week
ON YOUR BEHALF!

CALL: 8am - 10 pm  CST Monday - Saturday
1-800-811-2141 Code:46948 or call 1-800-226-0633 Code # 46948
Fax On Demand:  415-273-6020 (7 pgs.)
Outside the USA and Canada call:
1-785-762-6715 Code:46948 or 1-785-762-2442  Code # 46948

Get your own CODE# from the friendly operators and I'll
RACE YOU TO THE BANK! DON'T DELAY! MAKE THAT CALL!
I promise you'll be glad you did!
                         Al

********* Visit our website at: http://www.topsecrets100.com *********
                                           Code: 46948

P.S. After you come on board leave me a message at: 317-722-7707




From rem-conf Sat Jan 09 10:13:42 1999 
From rem-conf-request@es.net Sat Jan 09 10:13:41 1999
Received: from list by mail1.es.net with local (Exim 1.81 #2)
	id 0zz2jQ-0005bE-00; Sat, 9 Jan 1999 10:03:28 -0800
Received: from host-209-214-61-141.gso.bellsouth.net (localhost) [209.214.61.141] 
	by mail1.es.net with smtp (Exim 1.81 #2)
	id 0zz2jN-0005au-00; Sat, 9 Jan 1999 10:03:25 -0800
From: <cardacct@bellsouth.net>
Subject: ADV:CREDIT CARD PROCESSING
Date: Sat, 9 Jan 1999 06:20:10
Message-Id: <E0zz2jN-0005au-00@mail1.es.net>
Bcc:
To: rem-conf@es.net
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X-Loop: rem-conf@es.net
Precedence: list

*********************************************************
This message is being brought to you by World Teknologies
To be removed from from further mailings respond to this
message with "remove" in the subject line.
*********************************************************

Dear Friend,

Discover how you can accept credit cards directly
>from your website, telephone or fax for your products 
and services and never need to purchase or lease
expensive credit card equipment or pay a large monthly 
fee for online ordering capabilities or real time processing
transactions.

**Brand New**  Inteli-Charge Credit Card acceptance program 
allows you to accept Visa, MastercardTM, Amex and Discover 
any TIME,any WHERE through phone, fax or internet without 
the need to purchase or lease expensive credit card equipment. 
This brand new program will allow you to accept credit cards 
in 24-48 hours after submitting your application.

You simply pick up your telephone, dial a special toll free
800# 24 hrs a day 7 days a week, input a passcode and the
credit card # and receive an immediate authorization over the 
phone.
Within 2 days the money is deposited in your bank account. This 
is an exciting program for all businesses. Before you spend any 
money on a credit card merchant program LOOK at this new program! 
We have virtually a 100% approval for most business types 
regardless of past credit history!


If you have an interest in learning more about a Merchant Account 
for yourself or your business please email your Name, PHONE 
NUMBER (Don't forget your area code) and best time to call to:

mailto:cardacct@bellsouth.net
 
A representative will return your call within 24hrs.

Or feel free to call us on our 24 hour voicemail at:

1-800-600-0343 Ext:2104

P.S. ALSO AS PART OF A SPECIAL NATIONAL PROMOTION, IF
YOU ACT NOW YOU WILL RECEIVE A FREE ONLINE 
STORE WITH SECURE ORDERING CAPABILITIES
TO SELL YOUR PRODUCTS ONLINE! 

This offer only applies to U.S. Residents only and some Canadians

World Teknologies Inc.
7210 Jordan Ave.
Canoga Park Ca. 91303

 
 
 
 
 
 
 
 
 
 
 
 



From rem-conf Mon Jan 11 16:16:12 1999 
From rem-conf-request@es.net Mon Jan 11 16:16:11 1999
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	id 0zzrMc-0003j0-00; Mon, 11 Jan 1999 16:07:18 -0800
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	by mail1.es.net with esmtp (Exim 1.81 #2)
	id 0zzrMc-0003iq-00; Mon, 11 Jan 1999 16:07:18 -0800
Received: (from mperry@localhost)
	by george.lbl.gov (8.8.8/8.8.8) id QAA28202;
	Mon, 11 Jan 1999 16:06:26 -0800 (PST)
Date: Mon, 11 Jan 1999 16:06:26 -0800 (PST)
From: Marcia Perry (ITG staff) <mperry@george.lbl.gov>
Message-Id: <199901120006.QAA28202@george.lbl.gov>
To: DAAgarwal@lbl.gov, Jason_Sylvester@3com.com, Ksiple@home.com,
        MPerry@lbl.gov, M_Howard-Jeweler@lbl.gov, Rowe@BMRC.Berkeley.edu,
        bjorn@it.kth.se, cpmcparland@lbl.gov,
        ferenc@deepthought.EECS.Berkeley.EDU, ikeda@hike.te.chiba-u.ac.jp,
        itf@mcs.anl.gov, jam@mattheij.nl, jim.myers@pnl.gov, mbone@isi.edu,
        olson@mcs.anl.gov, philippe.galvez@cern.ch, rem-conf@es.net,
        roland@irdu.nus.edu.sg, toonen@mcs.anl.gov, van@ee.lbl.gov
Subject: Multithreaded devserv
X-Mailing-List: <rem-conf@es.net> 
X-Loop: rem-conf@es.net
Precedence: list

Hi All,
The Unix versions of devserv have been multithreaded (to
eliminate polling for receiving data over network sockets
and to speed up device startup/powering off).  A multi-
threaded Windows version is planned for a future release.
The updated binaries and source packages are available at
ftp://george.lbl.gov/mbone/devserv; please look for the
packages that contain "v1.0" in their titles.

Please let me know if there are any problems and feel
free to email feedback at any time.

Marcia Perry
_____________________________________________________________
Lawrence Berkeley National Lab
1 Cyclotron Road
Berkeley, CA 94720
MPerry@lbl.gov
Work Phone: (510) 486-6786
FAX #: (510) 486-6363
MS: 50B-2239



From rem-conf Tue Jan 12 02:17:09 1999 
From rem-conf-request@es.net Tue Jan 12 02:17:08 1999
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	id 1000nF-00003k-00; Tue, 12 Jan 1999 02:11:25 -0800
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	by mail1.es.net with esmtp (Exim 1.81 #2)
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	by pi4.informatik.uni-mannheim.de (8.8.8/8.8.8) with SMTP id LAA19829;
	Tue, 12 Jan 1999 11:11:19 +0100 (MET)
Date: Tue, 12 Jan 1999 11:11:19 +0100 (MET)
From: Christoph Kuhmuench <cjk@pi4.informatik.uni-mannheim.de>
X-Sender: cjk@pi4
To: rem-conf@es.net
cc: Gerald Kuehne <kuehne@pi4.informatik.uni-mannheim.de>
Subject: Payload for MPEG 4 ES stream?
Message-ID: <Pine.OSF.3.95.990112104603.11999B-100000@pi4>
MIME-Version: 1.0
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Hello,

we have got some questions concerning the AVT minutes: 

> A number of alternatives for the transport of MPEG-4 streams in an IP
> network were considered: 
>         - directly on UDP
>         - RTP followed by a full MPEG-4 SyncLayer packet
>         - MPEG-4 SyncLayer packets mapped onto RTP packets
>         - MPEG-4 elementary streams over RTP with natural payload
>           formats
> The preferred approach would be to use the latter approach, but since
> the ES interface is not a normative part of MPEG-4 this is not possible. 
>
> The approach chosen is, therefore, to map the MPEG-4 SyncLayer packets
> onto RTP packets, such that the common pieces of the header reside in
> the RTP header, with a small payload header providing the MPEG-4
> specific features. A single payload format is therefore used for MPEG-4
> streams transported within RTP, and the MPEG-4 model is maintained
> (although not the precise packet format).

Are there any ideas for compensating paket loss in the RTP/SL paket
scenario? Due to the lack of specific information about the media type in
the paket, the receiver can not easily recover from paket loss by using
RTP header information. 

Therefore, wouldn't it be useful to define at least some MPEG-4 ES
payloads for use in applications that do not use all features of MPEG-4.
Is there any work in progress or is the RTP payload for MPEG-4 SyncLayer
pakets the way to go?

Bye 
  Gerald Kuehne
  Christoph Kuhmuench




From rem-conf Tue Jan 12 07:14:32 1999 
From rem-conf-request@es.net Tue Jan 12 07:14:32 1999
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	id 1005Qr-0002mX-00; Tue, 12 Jan 1999 07:08:37 -0800
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	by mail1.es.net with smtp (Exim 1.81 #2)
	id 1005Qo-0002mM-00; Tue, 12 Jan 1999 07:08:35 -0800
From:     JOKEFARM <url-jokefarm@usa.net>
To:        <rem-conf@es.net>
Message-Id: <419.436172.42554028url-jokefarm@usa.net>
Subject: FREE PG13 JOKES DAILY-- JOIN OUR MAILING LIST IT'S FREE FREE
Mime-Version: 1.0
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Date: Tue, 12 Jan 1999 07:08:35 -0800
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JOKE FARMS =6 JOKES a week for FREE-FREE-FREE-FREE 
*************************************************************************************************
We do place  URL's in each mornings  mailing if you wish to have your's added just 
let us know at  URL-JOKEFARM-DATA@USA.NET   for more info with(URL) in the 
SUBJECT LINE. To PLACE a URL the cost is $15.00 per week for BUSINESS 
PAGES & $6.OO FOR  HOMEPAGES For more information ask us at the above 
address on how to add your URL.*****************************************************
*********************************************************************************************
NO PORNOGRAPHIC SITES OR MAILINGS, JUST GREAT CLEAN FUN.

PG-13 ALL THE WAY !!!!!!!!!!!!

To stay on the list : Place (YEP) in the suject area of < url-jokefarm-data@usa.net >
_____________________________________________________________________

ON BOTTOM OF PAGE you can be removed by a diffrent address.


JOKE-FARM JOKE OF THE DAY:

Remember Stella Liebeck, who pried $640,000 out of McDonalds after 
spilling a cup of McCoffee she was balancing between her knees in her 
grandson's car. 
But did you hear about John Parker, who tried a similar between-the-
legs McDonald's lawsuit - With a milkshake? (He lost). 
A guy sued Anheuser-Busch for emotional distress after drinking the 
company's beer without later having 'success with women.' 
A robber sued the bank, the manufacturer, the city, the police, and the 
hospital after a 'Security Pac' hidden in the stolen loot released tear gas 
and red dye. 
A minister and his wife sued a guide-dog school when a blind man 
stepped on the wife's toes in a shopping mall. 
A San Diego man sued the city for emotional trauma during a convert 
when he saw women using the men's rest room. 
A New York woman sued the company that makes The Clapper, 
claiming she 'had to clap too hard in order to turn her appliances on.' 
In a high-profile case last year, a Blue Cross-Blue Shield of Missouri 
worker unsuccessfully sued IBM, claiming that the 'faulty design' of 
IBM's keyboards produced pain that prevented her from working. 
In December 1996, a New York jury awarded $5.3 million to a secretary who 
claimed she was injured by a Digital Equipment Corporations keyboard. Rather 
than asserting that keyboards were defective, her lawyers contended that Digital 
neglected to warn its users about the dangers of typing.
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WE TAKE VISA MASTERCARD & DISCOVER CARDS
THOUSANDS WILL SEE IT: CALL US TOLL FREE AT 1-888-746-3311
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www.intellicast.com   Todays weather all over the world!!

www.emergency.com  Really neat to look at!!

www.nasdaq.com  Over the counter stocks!!

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www.dogpile.com  A whole lot of searching in one box !

www.rbc.net  RBC

www.weddingchannel.com  The wedding channel.



To be removed please place (DELETE) in SUBJECT.
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Thanks! Hope everyone has the day they deserve and have worked for & your 
dreams come true. 





From rem-conf Tue Jan 12 09:41:41 1999 
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Berkeley Multimedia, Interfaces and Graphics Seminar

Simplifying the Controls of an Interactive Movie Game 

(Wednesday January 20, 1999 12:30-2:00 PDT 405 Soda Hall) 

Jeff Johnson 
UI Wizards, Inc. 

Eight months before an interactive movie game was due to be shipped, its
developers and funders decided that its user interface had to be radically
simplified. The author was given the task of designing a new, simpler
control scheme. This paper describes the redesign process, the design
issues that arose and how they were resolved, the tests that were conducted
to evaluate new design ideas, and concludes with an evaluation of the
resulting design, lessons learned, and thoughts on user-interface design
vs. game design. 

Reference: 
Johnson, J., Simplifying the Controls of an Interactive Movie Game,
Proceedings of ACM CHI'98, pages 65-72. 

The seminar is broadcast on the Internet Mbone. The addresses are video
224.2.163.7/57076 and audio 224.2.147.61/27202. 

Sponsored by the Berkeley Multimedia Research Center

____________________________________________

Florissa Colina
Berkeley Multimedia Research Center (BMRC)
http://bmrc.berkeley.edu/
626 Soda Hall #1776
University of California
Berkeley, CA 94720-1776
Phone: (510) 643-0800   FAX: (510) 642-5615  
Email: florissac@bmrc.berkeley.edu



From rem-conf Tue Jan 12 14:15:50 1999 
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Date: Tue, 12 Jan 1999 17:34:06 -0500 (EST)
From: lezah <Lady_Lezah@usa.net>
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You are receiving this because you are on a Safe to e-mail list. Or someone sent me your address.If this 
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From rem-conf Wed Jan 13 00:47:11 1999 
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From: Stadler Wilhelm <wilhelm.stadler@infonova.at>
To: rem-conf@es.net
Cc: jmf-interest@Sun.COM
Subject: MPEG-2  Players capable of network streams?
Date: Wed, 13 Jan 1999 09:31:06 +0100
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Hi !

Does anybody know of any (free or commercial) implementation of

* A Windows-based MPEG-2 player (using SW or HW decoding based on
DirectShow)
	 that is capable of receiving MPEG-2 TS from a network
connection

	- over an ATM connection (MPEG-2 over AAL5, CBR, push-mode) 

	- over an IP connection (MPEG-2 over RTP/UDP/IP, pull or push)


	implemented as DirectShow plugin or JMF add-in (or something
else...)

Any tips appreciated!

TiA

Willi

-------------------------
DI Wilhelm Stadler
Head of Dept. Multimedia Systems
INFONOVA Information Technology
Karlauerguertel 1, A-8020 Graz
Tel: ++43 316 715440-0
Fax: ++43 316 715440-2
mailto:wilhelm.stadler@infonova.at





From rem-conf Wed Jan 13 01:26:02 1999 
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Subject: Re: MPEG-2  Players capable of network streams?
In-Reply-To: <573DE5317133D2119638006094B95B68212B2D@mailsrv.infonova.at> from Stadler Wilhelm at "Jan 13, 1999  9:31: 6 am"
To: wilhelm.stadler@infonova.at (Stadler Wilhelm)
Date: Wed, 13 Jan 1999 04:24:22 -0500 (EST)
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How about winamp. IT appears to be able to do this. IT does not currently
have multicasting extentions.

Stadler Wilhelm said:
> Hi !
> 
> Does anybody know of any (free or commercial) implementation of
> 
> * A Windows-based MPEG-2 player (using SW or HW decoding based on
> DirectShow)
> 	 that is capable of receiving MPEG-2 TS from a network
> connection
> 
> 	- over an ATM connection (MPEG-2 over AAL5, CBR, push-mode) 
> 
> 	- over an IP connection (MPEG-2 over RTP/UDP/IP, pull or push)
> 
> 
> 	implemented as DirectShow plugin or JMF add-in (or something
> else...)
> 
> Any tips appreciated!
> 
> TiA
> 
> Willi
> 
> -------------------------
> DI Wilhelm Stadler
> Head of Dept. Multimedia Systems
> INFONOVA Information Technology
> Karlauerguertel 1, A-8020 Graz
> Tel: ++43 316 715440-0
> Fax: ++43 316 715440-2
> mailto:wilhelm.stadler@infonova.at
> 
> 
> 




From rem-conf Wed Jan 13 01:31:07 1999 
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        jmf-interest@Sun.COM
Subject: Re: MPEG-2 Players capable of network streams? 
In-reply-to: Your message of "Wed, 13 Jan 1999 04:24:22 EST."
             <QQfxyr04891.199901130924@neserve0.uu.net> 
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Date: Wed, 13 Jan 1999 10:30:23 +0100
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I think there was some video involved here as well :-) 

/P




From rem-conf Wed Jan 13 02:05:28 1999 
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From: Stadler Wilhelm <wilhelm.stadler@infonova.at>
To: jhall@UU.NET
Cc: rem-conf@es.net, jmf-interest@Sun.COM
Subject: RE: MPEG-2  Players capable of network streams?
Date: Wed, 13 Jan 1999 10:56:41 +0100
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I mean MPEG-2 video MP@ML which is encoded/multiplexed in hardware and
played out by a special ATM NIC on  server, you obviously mean MPEG-2
Layer3 audio, no?
Regards,
Willi


> -----Original Message-----
> From: jhall@UU.NET [mailto:jhall@UU.NET]
> Sent: Wednesday, January 13, 1999 10:24 AM
> To: wilhelm.stadler@infonova.at
> Cc: rem-conf@es.net; jmf-interest@Sun.COM
> Subject: Re: MPEG-2 Players capable of network streams?
> 
> 
> How about winamp. IT appears to be able to do this. IT does 
> not currently
> have multicasting extentions.
> 
> Stadler Wilhelm said:
> > Hi !
> > 
> > Does anybody know of any (free or commercial) implementation of
> > 
> > * A Windows-based MPEG-2 player (using SW or HW decoding based on
> > DirectShow)
> > 	 that is capable of receiving MPEG-2 TS from a network
> > connection
> > 
> > 	- over an ATM connection (MPEG-2 over AAL5, CBR, push-mode) 
> > 
> > 	- over an IP connection (MPEG-2 over RTP/UDP/IP, pull or push)
> > 
> > 
> > 	implemented as DirectShow plugin or JMF add-in (or something
> > else...)
> > 
> > Any tips appreciated!
> > 
> > TiA
> > 
> > Willi
> > 
> > -------------------------
> > DI Wilhelm Stadler
> > Head of Dept. Multimedia Systems
> > INFONOVA Information Technology
> > Karlauerguertel 1, A-8020 Graz
> > Tel: ++43 316 715440-0
> > Fax: ++43 316 715440-2
> > mailto:wilhelm.stadler@infonova.at
> > 
> > 
> > 
> 



From rem-conf Wed Jan 13 06:35:27 1999 
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Subject: Re: MPEG-2  Players capable of network streams?
In-Reply-To: <573DE5317133D2119638006094B95B68212B43@mailsrv.infonova.at> from Stadler Wilhelm at "Jan 13, 1999 10:56:41 am"
To: wilhelm.stadler@infonova.at (Stadler Wilhelm)
Date: Wed, 13 Jan 1999 08:45:50 -0500 (EST)
Cc: jhall@UU.NET, rem-conf@es.net, jmf-interest@Sun.COM
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approve

yes that is what I mean. hehehe

Stadler Wilhelm said:
> I mean MPEG-2 video MP@ML which is encoded/multiplexed in hardware and
> played out by a special ATM NIC on  server, you obviously mean MPEG-2
> Layer3 audio, no?
> Regards,
> Willi
> 
> 
> > -----Original Message-----
> > From: jhall@UU.NET [mailto:jhall@UU.NET]
> > Sent: Wednesday, January 13, 1999 10:24 AM
> > To: wilhelm.stadler@infonova.at
> > Cc: rem-conf@es.net; jmf-interest@Sun.COM
> > Subject: Re: MPEG-2 Players capable of network streams?
> > 
> > 
> > How about winamp. IT appears to be able to do this. IT does 
> > not currently
> > have multicasting extentions.
> > 
> > Stadler Wilhelm said:
> > > Hi !
> > > 
> > > Does anybody know of any (free or commercial) implementation of
> > > 
> > > * A Windows-based MPEG-2 player (using SW or HW decoding based on
> > > DirectShow)
> > > 	 that is capable of receiving MPEG-2 TS from a network
> > > connection
> > > 
> > > 	- over an ATM connection (MPEG-2 over AAL5, CBR, push-mode) 
> > > 
> > > 	- over an IP connection (MPEG-2 over RTP/UDP/IP, pull or push)
> > > 
> > > 
> > > 	implemented as DirectShow plugin or JMF add-in (or something
> > > else...)
> > > 
> > > Any tips appreciated!
> > > 
> > > TiA
> > > 
> > > Willi
> > > 
> > > -------------------------
> > > DI Wilhelm Stadler
> > > Head of Dept. Multimedia Systems
> > > INFONOVA Information Technology
> > > Karlauerguertel 1, A-8020 Graz
> > > Tel: ++43 316 715440-0
> > > Fax: ++43 316 715440-2
> > > mailto:wilhelm.stadler@infonova.at
> > > 
> > > 
> > > 
> > 
> 




From rem-conf Wed Jan 13 06:49:11 1999 
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Subject: Re: MPEG-2 Players capable of network streams?
In-Reply-To: <199901130930.KAA15308@basil.cdt.luth.se> from Peter Parnes at "Jan 13, 1999 10:30:23 am"
To: peppar@cdt.luth.se (Peter Parnes)
Date: Wed, 13 Jan 1999 08:49:04 -0500 (EST)
Cc: jhall@UU.NET, wilhelm.stadler@infonova.at, rem-conf@es.net,
        jmf-interest@Sun.COM
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approve

well you know me, I'm not into video much. :-)

Peter Parnes said:
> I think there was some video involved here as well :-) 
> 
> /P
> 
> 




From rem-conf Thu Jan 14 15:57:27 1999 
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Date: Thu, 14 Jan 1999 15:47:08 -0800
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From: Florissa Colina <florissac@bmrc.berkeley.edu>
Subject: Reminder 1/20 Berkeley Multimedia, Interfaces and Graphics
  Seminar
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Berkeley Multimedia, Interfaces and Graphics Seminar

Simplifying the Controls of an Interactive Movie Game 

(Wednesday January 20, 1999 12:30-2:00 PDT 405 Soda Hall) 

Jeff Johnson 
UI Wizards, Inc. 

Eight months before an interactive movie game was due to be shipped, its
developers and funders decided that its user interface had to be radically
simplified. The author was given the task of designing a new, simpler
control scheme. This paper describes the redesign process, the design
issues that arose and how they were resolved, the tests that were conducted
to evaluate new design ideas, and concludes with an evaluation of the
resulting design, lessons learned, and thoughts on user-interface design
vs. game design. 

Reference: 
Johnson, J., Simplifying the Controls of an Interactive Movie Game,
Proceedings of ACM CHI'98, pages 65-72. 

The seminar is broadcast on the Internet Mbone. The addresses are video
224.2.163.7/57076 and audio 224.2.147.61/27202. 

Sponsored by the Berkeley Multimedia Research Center

____________________________________________

Florissa Colina
Berkeley Multimedia Research Center (BMRC)
http://bmrc.berkeley.edu/
626 Soda Hall #1776
University of California
Berkeley, CA 94720-1776
Phone: (510) 643-0800   FAX: (510) 642-5615  
Email: florissac@bmrc.berkeley.edu



From rem-conf Thu Jan 14 18:12:41 1999 
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Message-ID: <369EA2EF.37D7DADB@cs.columbia.edu>
Date: Thu, 14 Jan 1999 21:07:43 -0500
From: Henning Schulzrinne <hgs@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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Has anybody compiled vic for the Osprey 150 on Solaris 2.6? The Osprey
1500 version just shows a green local video image; the plain vanilla
version crashes. Rendezvous works, btw.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



From rem-conf Thu Jan 14 20:51:50 1999 
From rem-conf-request@es.net Thu Jan 14 20:51:49 1999
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	Fri, 15 Jan 1999 05:45:35 +0100 (MET)
Date: Fri, 15 Jan 1999 05:45:35 +0100 (MET)
From: Hans Mayer <Hans.Mayer@gmd.de>
Message-Id: <199901150445.FAA08542@mail.gmd.de>
To: hgs@cs.columbia.edu, rem-conf@es.net
Subject: Re:  vic with Osprey 150?
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>Has anybody compiled vic for the Osprey 150 on Solaris 2.6? The Osprey
>1500 version just shows a green local video image; the plain vanilla
>version crashes. Rendezvous works, btw.

I'm not shure about the 150. There is a version from University of
Erlangen that works (more or less) with the 1100 and 1500 (I'm having
problems with the initialisation of the card - default is NTSC and
I have to start o1k_display to init the card to PAL but this program
hangs (in 99%) my U30).

Kind Regards,
Hans Mayer, GMD - TKT



From rem-conf Thu Jan 14 23:18:17 1999 
From rem-conf-request@es.net Thu Jan 14 23:18:16 1999
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	Fri, 15 Jan 1999 09:14:35 +0200 (IST)
Message-ID: <369EEAEB.43ED7092@internet-zahav.net>
Date: Fri, 15 Jan 1999 09:14:52 +0200
From: PUIU BIRNBAUM <pamona@internet-zahav.net>
X-Mailer: Mozilla 4.5 [en] (Win95; I)
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To: rem-conf@es.net
Subject: IMPORT EXPORT SERVICES
Content-Type: multipart/mixed;
 boundary="------------2C3B4A900B7DF220EB5B8C55"
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This is a multi-part message in MIME format.
--------------2C3B4A900B7DF220EB5B8C55
Content-Type: text/plain; charset=us-ascii
Content-Transfer-Encoding: 7bit

PAMONA INTERNATIONAL
    Import and Export Consulting.

You need a good buyer for the goods you are exporting.
You need a good manufacturer for the goods you are importing.
" A good partner is the best business "
This is what we offer you : A GOOD PARTNER.
We will bring you the best partners to your desk.
We have all the tools and 35 years of actual practice in import,
export, consulting, international transport, banking, financing,
and all the other aspects of the international trade.
We offer you our services to solve the problems for which you
don't have the time, or connections, or information.
Tell us what you need and we will take care of it, against real
reasonable agreed charges.

Will be pleased to meet you,

Puiu Birnbaum
Dipl. M.A. Economist.


PAMONA INTERNATIONAL
21/6 Miler Street
Rehovot - 76284 Israel
Phone :++972-8-9462120   Fax :++972-8-9471164
E-Mail : pamona@internet-zahav.net
Web site : http://members.xoom.com/Pamona/partner.html

--------------2C3B4A900B7DF220EB5B8C55
Content-Type: text/html; charset=us-ascii;
 name="partner.html"
Content-Transfer-Encoding: 7bit
Content-Disposition: inline;
 filename="partner.html"

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<head>
   <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1">
   <meta name="GENERATOR" content="Mozilla/4.5 [en] (Win95; I) [Netscape]">
   <title>pamona</title>
</head>
<body text="#000000" link="#FF0033" vlink="#660066" alink="#660066" background="teal_paper.gif">

<center>
<h1>
<u><font size=+2>PAMONA INTERNATIONAL</font></u><img SRC="y1.JPG" BORDER=0 height=240 width=320></h1></center>

<center>
<h1>
<font size=+2>Import and Export Consulting</font></h1></center>
<font size=+2>&nbsp;"A good Partner is the best Business". You need a good
buyer for the goods you are exporting. You need a good manufacturer for
the goods you are importing. This is what we offer you : A GOOD PARTNER.
We will bring you the best partners to your desk. We have all the tools
and 35 years of actual practice in import, export, consulting, international
transport, banking, financing, and all the other aspects of the international
business. Tell us what you need and we will take care of it !</font>
<p><font size=+2>Will be pleased to meet you,</font>
<p><font size=+2>P. Birnbaum</font>
<br><font size=+2>Dipl. M.A. Economist.</font>
<br><img SRC="fb016.jpg" BORDER=0 height=180 width=240> <font size=+2>Please
contact us :</font>
<br><font size=+2>PAMONA INTERNATIONAL</font>
<br><font size=+2>21/6 Miler Street</font>
<br><font size=+2>Rehovot - 76284 Israel</font>
<br><font size=+2>Phone : ++972-8-9462120&nbsp;&nbsp;&nbsp; Fax : ++972-8-9471164</font>
<p><font size=+2>E-Mail :</font>
<br><i><font size=+2><a href="mailto:pamona@netscape.net">pamona@netscape.net</a></font></i>
<br>&nbsp;
<br>&nbsp;
<p><font size=+1>.</font>
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
</body>
</html>


--------------2C3B4A900B7DF220EB5B8C55--




From rem-conf Fri Jan 15 00:00:26 1999 
From rem-conf-request@es.net Fri Jan 15 00:00:25 1999
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	id 10148k-0001OC-00; Thu, 14 Jan 1999 23:57:58 -0800
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Message-Id: <v03010d50b2c459b87fa9@[internet.email]>
Mime-Version: 1.0
Content-Type: text/plain; charset="us-ascii"
Approved-by: Tempting Tear-Outs for general release
X-Priority: 1 (Highest)
Date: Thu, 14 Jan 1999 22:44:31 -0400
To: Tempting Tear-Outs <temptingtearouts@1stconnect.com>
From: Tempting Tear-Outs <temptingtearouts@1stconnect.com>
Subject: Hi
X-Mailing-List: <rem-conf@es.net> 
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===>> FREE 1 yr USA Magazine Sub sent worldwide-200+ Choices!  Up to $81.00
value!

To be removed from our mailing list, please send email to:
temptingtearouts@1stconnect.com with the subject line of "remove."


FOR MORE INFO:   please "cut out" the below form on the "cut" lines shown,
and fax it, for the fastest reply to:                1-718-227-9125   (this
is a fax # in the USA)

or send via smail (first class mail or airmail) to:
                                         Tempting Tear-Outs
                                         Att. Free-catalogue-by-email Dept
                                         3835 Richmond Ave.  Suite #200
                                         Staten Island NY  10312-3828
                                         USA

SORRY, BUT.... our software is not set up to accept the below form via
return email;   WE CAN ONLY acknowledge forms sent in via fax or smail.

--> IMPORTANT complete directions, to ensure that you get a reply, and more
info follow, below the reply form and the catalogue options.


*------------cut here/begin-------------------------------------------*

Name (First Middle Last):
Internet email address:
Smail home address:
City-State-Zip:
Country:
Work Tel. #:
Work Fax #:
Home Tel. #:
Home Fax #:
Cellular (Mobile) Tel. #:
Beeper (Pager) Tel. #:

How did you hear about us (name of person/company who referred you or the
area of
the internet that you saw us mentioned in):  Referred by:  Tempting
Tear-Outs
011499-l-hi

Name of USA mags you currently get on the newsstand or in the store:

Name of USA mags you currently get on a subscription basis, through the mail:

Name of USA mags you would like price quotes on when we call you:

Catalogue version desired (list number of choice below):

*------------cut here/end--------------------------------------------*



CATALOGUE VERSION CHOICES:

1.  This version can be read by everyone, no matter what type of
     computer you use, or what type of software you use.  It is a simple
     format, with just our entire catalogue pasted into the body of a
     single email message, 316K in size.  If you use pine or elm on a unix
     system or an advanced software version such as Eudora Pro 3.0 or
     later, you will most likely receive it as a single email message.
     However, if your software limits incoming email messages to a
     certain size, say 32K or so, then your software will split it into
     multiple email message parts.   Whether you receive it as a single
     email message or multiple part email messages, you can easily
     paste it into one whole text document with your word processor, in
     about 10 minutes or so.
2.  For more advanced computer users:  attached plain ascii text file
     ~316K - you must know how to download an attached text file and
     then be able to locate it on your hard drive or system home
     directory;  it can then be opened with any pc or mac word processing
     software.  If in doubt, don't ask for this version.  This isn't for
     internet *newbies.* Better to order option 1 and spend a few minutes
     pasting them into one whole text document with your word processor,
     than to waste hours trying to figure how to deal with this option.
     This version is great for doing keyword searches and jumping around
     within the catalogue with your word processing software, if your
     normal email reading software doesn't allow this.


VERY IMPORTANT DIRECTIONS TO ENSURE THAT YOU GET A REPLY:

1.   you must call from an "unblocked number," ie. one that is not blocked
>from caller id.  We are very sorry for this requirement, but our fax
software requires this before it allows an incoming fax call to connect.
If you have a blocked number, you must first unblock it.  In most cases
this means dialing *82 from a touch-tone phone (or 1182 from a rotary
phone) before you dial 1-718-227-9125.     NOTE:  If you are not sure if
your number is blocked, just try dialing our fax # normally.  If you don't
get a recording telling you your number is blocked, your number has been
transmitted and you may press the start button on your fax when you hear
the fax tone from our fax.
2.   no reply forms can be accepted by email....only via fax or smail.
3.   your form must be typewritten or printed out on your computer printer
before you fax it;  sorry, but *no* handwritten forms will be acknowledged.
If you can't find someone with a typewriter or a computer printer, we
apologize for not being able to reply to you.
4.   faxes with cover pages will be rejected.  You must send *only* the
reply form.
5.   forms not *completely* filled in will not be acknowledged.
6.   you will receive a reply within 1 business day directly from the
company making the offer via email.  Therefore you must have an email
address.  If you read this message, then you must have an email address, or
access to one, at least.   :-)
7.   your fax must not exceed 1 page in length.   Faxes of 2 or more pages
will be sensed, then auto-terminated and deleted.  Your fax goes directly
onto our 5.0 gigabyte hard drive and we must limit all incoming faxes to 1
page.
8.   all faxes must begin with:
*------------cut here/begin-------------------------------------------*
and must end with:
*------------cut here/end--------------------------------------------*
9. Any fax not conforming to this format will be sensed by our software,
then auto-terminated and deleted from the hard drive, before any human ever
gets to see it.
10. The type on your fax must be dark and legible.   If in doubt, please
print it out darker before faxing it in.  If we can't read it, we can't
reply to you or send you our FREE catalogue.     :-(
11.  If this all seems too complicated for faxing, just do it the old
fashioned way via smail!!!


WHO WE ARE:

Tempting Tear-Outs is an advertising company that brings potential new
customers to the companies they advertise for.

MORE ABOUT THE COMPANY MAKING THE FREE OFFER:

The company making the offer is a magazine subscription agency based in the
USA.  They have over 1,100 popular USA titles available to be shipped to
*any* country, including of course, to anywhere in the USA!    They offer a
FREE 1 yr. subscription to your choice of over 200 of the titles in their
catalogue to any new customer using them for the first time.       The
dollar value of the freebies, based on the subscription prices directly
>from the publishers, ranges from $6.97 all the way up to $50.00!

For new customers in the USA, there is no charge for FPH (foreign postage &
handling), so the freebie is 100% free!   For new customers living
overseas, the only charge on the freebie would be for the FPH (foreign
postage & handling).

Their president has been in the magazine subscription business since 1973
and they are very customer-service oriented.   They will even help you with
address changes on your magazines, even if you move from one country to
another country.   They have thousands of happy customers in over 59
countries.

Their price guarantee is very simple:       they guarantee that their
subscription prices are the lowest available and they will BEAT any
legitimate, verifiable offer before you pay them or match it afterwards, by
refunding you the difference in price PLUS the cost of the postage stamp
you would use sending in the special offer to them, even 6 months after you
pay them, as long as it was current at the time of your offer.    Does that
sound fair?       Wouldn't it be great if everything you bought came with
that price guarantee?

Sometimes they are less than half of the next best deal out there,
sometimes just a little cheaper, but always you get the lowest rates
without having to shop around.     With 1,100+ titles on their list, they
would like to think that they have also the best selection around!

Within the USA, for their USA customers, they are cheaper than all their
competitors and even the publishers themselves.  This is their price
guarantee.         The 1 yr. freebie that you get with your first order is
completely free!

Overseas, (even after you factor in the cost of the FPH (foreign postage &
handling) and the conversion from USA Dollars to your currency), on the
average, they are generally around one-fourth to one-half of what the
newsstands overseas charge locally for USA magazines.  On some titles they
are as little as one-tenth of what the newsstands charge.  They are also
the cheapest subscription source for delivery overseas, including directly
>from the publishers themselves!   Some publishers don't even offer
subscriptions overseas.........but overseas subscriptions are this
company's specialty!  They feel that magazines should not be a luxury
overseas.   In the USA, people buy magazines and then toss them after
reading them for just a few minutes or hours.  They are so cheap in the
USA!   Well, this company would like to make it the same way for their
overseas customers.  They are also cheaper than all their competitors in
the USA and overseas, including the publishers themselves!   It is also
*highly unlikely* you will find any of their USA competitors calling you
overseas, in order to offer that personal touch, just to sell you a couple
of magazines!  But that is what this company specializes in and loves
doing!     Around one-half their business comes from overseas, so they are
very patient with new customers who only speak limited English as a 2nd
language.    Subscription prices quoted for overseas consist of the
subscription price, plus the FPH.   You add the two together and that is
your total cost.   The exception is the 1 yr. freebie you get with your
first order.   On that title, you pay *only* the FPH for the 1 yr. term.

Their prices are so cheap because when you deal with them, you cut-out all
the middlemen.


HERE IS HOW YOU CAN GET MORE INFO AND GET STARTED WITH THEM:

Simply fax or smail back to us the reply form listed at the top of this
message.   We will then forward your form on to the subscription agency.
They will then email their "big and juicy" catalogue to you, in whichever
of the two formats you chose.   The catalogue is FREE and makes for hours
of fascinating reading, on its own. It includes the complete list of
freebies, a complete list of all the titles they sell, as well as detailed
descriptions on most of the titles, along with lists of titles by category
of interest and their terms of sale.

They will then give you a friendly, no-pressure, no obligation, 5-minute
call to go over how they work and to answer any questions that you might
have, as well as give you up-to-the minute price quotes on any titles you
might be considering.     They will call you in whatever country you live
in, taking the time difference into account.        As they like to
emphasize the personal touch they give to each new customer, all first-time
orders can only be done via phone, so they can answer all your questions
completely and personally.   Once you have placed your first order via
phone, you will be able to place future orders and make inquiries on your
account, get price quotes, etc., all via email, if that is most convenient
for you.

Within the USA, they accept payment via check over the phone, Mastercard,
Visa, American Express, Diner's Club and Carte Blanche.    Overseas, they
accept Mastercard, Visa, American Express, Diner's Club and Carte Blanche,
even if your credit card is a local one in local currency (that most
merchants in the USA would not normally be willing to accept).

That's our introduction of our client that we represent.   We hope that we
have piqued your interest and that you will take the next step to get their
free catalogue!   Thank you for your time and interest.

--
Tempting Tear-Outs.
For more info on advertising rates, please write us on your company
letterhead, w/business card, via smail to:   Tempting Tear-Outs, 3835
Richmond Ave. Suite #200, Staten Island NY  10312-3828, USA.





From rem-conf Fri Jan 15 01:31:16 1999 
From rem-conf-request@es.net Fri Jan 15 01:31:15 1999
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	id 1015WL-0002j4-00; Fri, 15 Jan 1999 01:26:25 -0800
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          Fri, 15 Jan 1999 09:24:49 -0000
From: "Swale,RP,NAM5 SWALERP M" <richard.swale@bt.com>
To: rem-conf@es.net
Subject: remove
Date: Fri, 15 Jan 1999 09:21:16 -0000
X-Mailer: Internet Mail Service (5.5.2232.9)
Message-Id: <E1015WK-0002iG-00@mail1.es.net>
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remove


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M````P````````$8`````#H4````````#`!>`""`&``````#`````````1@``
M```8A0```````!X`)8`((`8``````,````````!&`````#:%```!`````0``
M```````>`":`""`&``````#`````````1@`````WA0```0````$`````````
M'@`G@`@@!@``````P````````$8`````.(4```$````!``````````,`)@``
M`````P`V```````>`#%``0````T````X,#(U,C$Q-3`P,#$``````P`:0```
M```>`#!``0````T````X,#(U,C$Q-3`P,#$``````P`90``````#`(`0____
M_P(!^3\!````7`````````#<IT#(P$(0&K2Y"``K+^&"`0````8````O3SU"
M5"]/53U21"U-05)43$532$%-+T-./5)%0TE0245.5%,O0TX]35,@34%)3"]#
M3CTX,#(U,C$Q-3`P,#$`'@#X/P$````8````4W=A;&4L4E`L3D%--2!35T%,
M15)0($T`'@`X0`$````-````.#`R-3(Q,34P,#`Q``````(!^S\!````7```
M``````#<IT#(P$(0&K2Y"``K+^&"`0````8````O3SU"5"]/53U21"U-05)4
M3$532$%-+T-./5)%0TE0245.5%,O0TX]35,@34%)3"]#3CTX,#(U,C$Q-3`P
M,#$`'@#Z/P$````8````4W=A;&4L4E`L3D%--2!35T%,15)0($T`'@`Y0`$`
M```-````.#`R-3(Q,34P,#`Q`````$``!S``ZRI>:$"^`4``"##POE[8:$"^
M`1X`/0`!`````0`````````>`!T.`0````<```!R96UO=F4```L`*0``````
M"P`C```````#``80%7NZ*0,`!Q`&`````P`0$``````#`!$0`````!X`"!`!
1````!P```%)%34]610``"+`=
`
end



From rem-conf Fri Jan 15 02:47:03 1999 
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To: Henning Schulzrinne <hgs@cs.columbia.edu>
Cc: rem-conf@es.net
Subject: Re: vic with Osprey 150? 
In-Reply-To: Message from Henning Schulzrinne <hgs@cs.columbia.edu> 
   of "Thu, 14 Jan 1999 21:07:43 EST." <369EA2EF.37D7DADB@cs.columbia.edu> 
Mime-Version: 1.0
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	boundary="==_Exmh_13771006000"
Date: Fri, 15 Jan 1999 11:42:47 +0100
From: Frank Lyonnet <Frank.Lyonnet@sophia.inria.fr>
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This is a multipart MIME message.

--==_Exmh_13771006000
Content-Type: text/plain; charset=us-ascii

> Has anybody compiled vic for the Osprey 150 on Solaris 2.6? The Osprey
> 1500 version just shows a green local video image; the plain vanilla
> version crashes. Rendezvous works, btw.

  There's some problems with the xil scaling procs under Solaris 2.6 and 
Osprey boards.
I'm not sure it comes from the chip itself or the xil revision shipped with 
solaris 2.6
(I had the same problems with an early osprey board that was ok under 2.5 and 
no longer under 2.6).
 
The reason why Rendezvous works is that I simply disable hardware scaling on 
such boards ...
If you want to re-enable it define the O1KSCALEBUG in your environnement.

Included is the source code of the xil driver for Rendez-vous.

Frank Lyonnet

Game And Internet Architecture - GAIA Project - INRIA Sophia Antipolis
http://gaia.inria.fr/

Rendez-Vous home page 
http://www.inria.fr/rodeo/rv/


--==_Exmh_13771006000
Content-Type: text/plain; name="videograber_sunvideo.cc"; charset=us-ascii
Content-Description: videograber_sunvideo.cc
Content-Transfer-Encoding: quoted-printable
Content-Disposition: attachment; filename="videograber_sunvideo.cc"

/* videograber_sunvideo.cc */

#include "videograber_sunvideo.h"
#include "videostdsize.h"
#include "mathutils.h"
#include <string.h>
#include <stdio.h>

video_graber_sunvideo::video_graber_sunvideo(int _Width, int _Height, col=
orspace _CSpace, digitizer_port _Port, digitizer_signal _Suggested, BOOLE=
AN _DoubleBuffering, BOOLEAN * Successfull) : video_graber_hard(_Width, _=
Height, _CSpace, _Port, _Suggested, _DoubleBuffering, Successfull)
{
#ifndef __SUNOS__

  Image =3D NULL;
  RTVCImage =3D NULL;
  ScaledImageA =3D NULL;
  ScaledImageB =3D NULL;

  /* Open device */
  *Successfull =3D open(NULL); =


  if(*Successfull)
  {
    /* Ok card is opened, now try to satisfy requested specs */
    request_secam();
    request_port();
    request_size();
    request_colorspace();

    /* And allocate image */
    allocate_image();

    *Successfull =3D (get_format() !=3D SIGNAL_NONE); =

  };

#else

  *Successfull =3D FALSE;

#endif
}

video_graber_sunvideo::~video_graber_sunvideo(void)
{
#ifndef __SUNOS__

  /* Close device */
  close();
#endif
}

void video_graber_sunvideo::change_colorspace(colorspace _CSpace)
{
#ifndef __SUNOS__

  video_graber::change_colorspace(_CSpace);

  destroy_image();
  request_colorspace();
  allocate_image();

#endif
}

void video_graber_sunvideo::change_size(int _Width, int _Height)
{
#ifndef __SUNOS__

  video_graber::change_size(_Width, _Height);

  destroy_image();
  request_size();
  allocate_image();

#endif
}

void video_graber_sunvideo::do_grab(void)
{
#ifndef __SUNOS__

  /* Swap buffers */
  if(DoubleBuffering)
    Image->swap_buffers();

  /* Scale */
  xil_scale(RTVCImage, ((video_sample_image_xil *)Image)->get_xil_image()=
, "nearest", XScale, YScale);
  xil_toss(((video_sample_image_xil *)Image)->get_xil_image());

#endif
}

digitizer_signal video_graber_sunvideo::get_format(void)
{
#ifndef __SUNOS__

char * Format =3D NULL;
digitizer_signal Ret;

  /* Description : Returns 0 if capturing from PAL, non-zero if capturing=
 from NTSC */
  xil_get_device_attribute(RTVCImage, "FORMAT_V", (void **) &Format);  =

  =

  if(Format =3D=3D NULL)
  {
    Ret =3D SIGNAL_PAL;
  }
  else
  {
    Ret =3D SIGNAL_NTSC;
  };

  /* Sunvideo cannot determine if signal is none, detect black image */
  {
  int i, j;
  XilMemoryStorage DataStore;
  video_sample * Ptr, * Ptr0;
  int PStr, LStr;

    /* Scale */
    xil_scale(RTVCImage, ScaledImageA, "nearest", XScale, YScale);
    xil_toss(ScaledImageA);

    xil_get_memory_storage(ScaledImageA, &DataStore);
    Ptr =3D Ptr0 =3D DataStore.byte.data;
    PStr =3D DataStore.byte.pixel_stride;
    LStr =3D DataStore.byte.scanline_stride;

    for(j =3D 0; j < Height; j ++)
    {
      for(i =3D 0; i < Width; i ++)
      {
        /* Well none signal is noisy but no sup of 100 */
        if(*Ptr > 100)
          goto retsignal;
         Ptr +=3D PStr;
      }
      Ptr0+=3DLStr;
    }
  }
  return(SIGNAL_NONE);

retsignal :

  return(Ret);

#else

  return(SIGNAL_NONE);

#endif
}

void video_graber_sunvideo::request_colorspace(void)
{
#ifndef __SUNOS__

  /* Card do only 444 YCBCR */

  if(CSpace =3D=3D COLORSPACE_COLOR)
    CSpace =3D COLORSPACE_YCBCR_444;

  if(CSpace =3D=3D COLORSPACE_GRAY)
    CSpace =3D COLORSPACE_YCCIR;

#endif
}


void video_graber_sunvideo::request_size(void)
{
#ifndef __SUNOS__
int RWidth, RHeight;
int NBands;
XilDataType DataType;
int DivFactor2;

  xil_get_info(RTVCImage, (u_int *)&RWidth, (u_int *)&RHeight, (u_int *) =
&NBands, &DataType);

  /* Only supported with SunVideo SBus */
  if((strcmp(DeviceName, "o1k") =3D=3D 0) && (getenv("O1KSCALEBUG") =3D=3D=
 NULL))
  {
    XScale =3D 1.0;
    YScale =3D 1.0;

    Width =3D RWidth;
    Height =3D RHeight;
 =

    return;
  }

  /* We got a probleme with NTSC cameras ... we cannot provide xCIF image=
s directly, we use */
  /* div2 larger format instead but it is way slower than PAL ... */

  /* The scaler is fast only with simple scales */
  /* Just use it, will stretch later */
  /* Allow an image to be 80 % of requested */

  #define ALLOWED_ROUND 0.8

  /* Select div 2 factor */
  DivFactor2 =3D 0;
  while(((RWidth / mypow2(DivFactor2 + 1)) >=3D (int)((float)Width * ALLO=
WED_ROUND)) && ((RHeight / mypow2(DivFactor2 + 1)) >=3D (int)((float)Heig=
ht * ALLOWED_ROUND)))
    DivFactor2 ++;

  if(DivFactor2 > 0)
  {
    XScale =3D 1.0 / mypow2(DivFactor2);
    YScale =3D 1.0 / mypow2(DivFactor2);

    Width =3D RWidth  / mypow2(DivFactor2);
    Height =3D RHeight / mypow2(DivFactor2);
  } =

  else
  {
    XScale =3D 1.0;
    YScale =3D 1.0;

    Width =3D RWidth;
    Height =3D RHeight;
  }
   =

#endif
}

void video_graber_sunvideo::request_secam(void)
{
#ifndef __SUNOS__

  /* To do */

#endif
}

void video_graber_sunvideo::request_port(void)
{
#ifndef __SUNOS__

int PortNumber;

  switch(Port)
  {
    case PORT_COMPOSITE1 :
      PortNumber =3D 0;
    break;
    case PORT_COMPOSITE2 :
      PortNumber =3D 1;
    break;
    case PORT_SVIDEO1 :
      PortNumber =3D 2;
    break;
    case PORT_SVIDEO2 :
      PortNumber =3D 2;
    break;
  };

  if(xil_set_device_attribute(RTVCImage, "PORT_V", (void *)((PortNumber +=
 1)%3)) =3D=3D XIL_FAILURE)
  {
    fatal_error("setting PORT_V attribute failed");
  }

#endif
}

void video_graber_sunvideo::allocate_image(void)
{
#ifndef __SUNOS__
int NBands;
int RWidth, RHeight;
XilDataType DataType;

  xil_get_info(RTVCImage, (u_int *)&RWidth, (u_int *)&RHeight, (u_int *) =
&NBands, &DataType);

  if((ScaledImageA =3D xil_create(State, Width, Height, NBands, XIL_BYTE)=
) =3D=3D NULL) =

  {
    fatal_error("xil_create scaled_image failed");
  };

  if(DoubleBuffering)
  {
    if((ScaledImageB =3D xil_create(State, Width, Height, NBands, XIL_BYT=
E)) =3D=3D NULL) =

    {
      fatal_error("xil_create scaled_image failed");
    };
  };

  Image =3D new video_sample_image_xil(Width, Height, CSpace, NULL, Doubl=
eBuffering, ScaledImageA, ScaledImageB);

#endif
}

void video_graber_sunvideo::destroy_image(void)
{
#ifndef __SUNOS__

  if(ScaledImageA !=3D NULL)
    xil_destroy(ScaledImageA);

  if(ScaledImageB !=3D NULL)
    xil_destroy(ScaledImageB);

  delete Image;

#endif
}

BOOLEAN video_graber_sunvideo::open(char * Name)
{
#ifndef __SUNOS__
char SunDevice[256];
XilDevice Device;
int RWidth, RHeight;
int NBands;
XilDataType DataType;
 =

  if(Name =3D=3D NULL)
  {
    if(system("\\ls /dev/o1k0 > /dev/null") =3D=3D 0)
      strcpy(DeviceName, "o1k");
    else
      strcpy(DeviceName, "rtvc");
  }
  =

  if(strcmp(DeviceName, "o1k") =3D=3D 0)
  {
    sprintf(SunDevice, "MMAC%s", DeviceName);
  }    =

  else
  {
    sprintf(SunDevice, "SUNW%s", DeviceName);
  }

  RTVCImagePrimary =3D RTVCImage =3D NULL;

  if((State =3D xil_open()) =3D=3D NULL) =

  {
    debug_trace(DEBUG_DEVICE, "can't open xil library");
    return(FALSE);
  }

  /* To turn on debugging put 1, else 0 */ =

  xil_state_set_show_action(State, 0);

  if(!(Device =3D xil_device_create(State, SunDevice))) =

  {
    debug_trace(DEBUG_DEVICE, "cannot create device object");
    return(FALSE);
  }

  RTVCImagePrimary =3D xil_create_from_device(State, SunDevice, Device);

  xil_device_destroy(Device);

  if(RTVCImagePrimary =3D=3D NULL)
  {
    debug_trace(DEBUG_DEVICE, "can't open SunVideo software driver");
    return(FALSE);
  }

  /* Here deal with the color ... */
  /* Only supported with SunVideo SBus */
  if(strcmp(DeviceName, "rtvc") =3D=3D 0)
  {
    xil_get_info(RTVCImagePrimary, (u_int *)&RWidth, (u_int *)&RHeight, (=
u_int *) &NBands, &DataType);
    if(CSpace =3D=3D COLORSPACE_GRAY)
      RTVCImage =3D xil_create_child(RTVCImagePrimary, 0, 0, RWidth, RHei=
ght, 0, 1);
    else
      RTVCImage =3D RTVCImagePrimary;
  }
  else
    RTVCImage =3D RTVCImagePrimary;

  return(TRUE);

#else

  return(FALSE);

#endif
}

void video_graber_sunvideo::close(void)
{
#ifndef __SUNOS__
  if(RTVCImage !=3D NULL)
    xil_destroy(RTVCImage);

  if((RTVCImagePrimary !=3D RTVCImage) && (RTVCImagePrimary !=3D NULL))
    xil_destroy(RTVCImagePrimary);

  if(State !=3D NULL)
    xil_close(State);
#endif
}


--==_Exmh_13771006000--





From rem-conf Fri Jan 15 09:00:47 1999 
From rem-conf-request@es.net Fri Jan 15 09:00:46 1999
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	by lima.irisa.fr (8.9.1a/8.9.1) with ESMTP id RAA18343;
	Fri, 15 Jan 1999 17:49:30 +0100 (MET)
Message-ID: <369F7169.2279AD19@irisa.fr>
Date: Fri, 15 Jan 1999 17:48:41 +0100
From: Christine Guillemot <Christine.Guillemot@irisa.fr>
X-Mailer: Mozilla 4.03 [fr] (WinNT; I)
MIME-Version: 1.0
To: cjk@pi4.informatik.uni-mannheim.de
CC: rem-conf@es.net, Paul.Christ@rus.uni-stuttgart.de,
        wesner@rus.uni-stuttgart.de, Christine.Guillemot@irisa.fr
Subject: Re: Payload for MPEG 4 ES ?
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Hello,

Chistoph Kuhmuench wrote:

> Hello,

> we have got some questions concerning the AVT minutes:

>> A number of alternatives for the transport of MPEG-4 streams in an IP

>> network were considered:
>>         - directly on UDP
>>         - RTP followed by a full MPEG-4 SyncLayer packet
>>         - MPEG-4 SyncLayer packets mapped onto RTP packets
>>         - MPEG-4 elementary streams over RTP with natural payload
>>           formats
>> The preferred approach would be to use the latter approach, but since

>> the ES interface is not a normative part of MPEG-4 this is not
possible.
>>
>> The approach chosen is, therefore, to map the MPEG-4 SyncLayer
packets
>> onto RTP packets, such that the common pieces of the header reside in

>> the RTP header, with a small payload header providing the MPEG-4
>> specific features. A single payload format is therefore used for
MPEG-4
>> streams transported within RTP, and the MPEG-4 model is maintained
>> (although not the precise packet format).

> Are there any ideas for compensating paket loss in the RTP/SL paket
> scenario? Due to the lack of specific information about the media type
in
> the paket, the receiver can not easily recover from paket loss by
using
> RTP header information.

> Therefore, wouldn't it be useful to define at least some MPEG-4 ES
> payloads for use in applications that do not use all features of
MPEG-4.
> Is there any work in progress or is the RTP payload for MPEG-4
SyncLayer
> pakets the way to go?

At INRIA, together with the university of Stuttgart, we are working
toward
a payload for transporting directly  MPEG-4 ES over RTP,
that would also support mechanisms for protection against packet loss.
A very preliminary draft has been presented at the IETF AVT Orlando
meeting, the
corresponding document, currently being amended/updated ... can be found
in the
IETF site under the name  draft-guillemot-genrtp-00.txt.

The implementation of this payload for transporting MPEG-4 ES over RTP,
for
validation purposes, is also in progress in the framework of a European
project.

Best Regards,

Christine Guillemot.


> Bye
>  Gerald Kuehne
>   Christoph Kuhmuench







From rem-conf Sat Jan 16 02:52:06 1999 
From rem-conf-request@es.net Sat Jan 16 02:52:04 1999
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Date: Sat, 16 Jan 1999 02:40:56 -0800
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From rem-conf Sat Jan 16 16:38:09 1999 
From rem-conf-request@es.net Sat Jan 16 16:38:08 1999
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From:     "Best Value of America, Inc." <Desktop13@hotmail.com>
To:        <rem-conf@es.net>
Message-Id: <419.436176.80931690Desktop13@hotmail.com>
Subject: Marketing Tools for Your Business - Free Software
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Date: Sat, 16 Jan 1999 16:32:44 -0800
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From rem-conf-request@es.net Mon Jan 18 07:42:28 1999
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From:     "Best Value of America, Inc." <Desktop@mygo.com>
To:        <rem-conf@es.net>
Message-Id: <419.436178.43115868Desktop@mygo.com>
Subject: Marketing Tools for Your Business - Free Software
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Date: Mon, 18 Jan 1999 07:28:13 -0800
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has changed all that forever!!  Desktop Server turns your PC into a "Real" Bulk 
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Our apologies if you have received multiple copies of this CPF. 

                              C A L L   F O R   P A P E R S
                              =============================
___   __   __     __  __  
 | | |  | |  | / |__||__|      International Symposium on 
 | | |  | |--|     /   /    DISTRIBUTED OBJECTS AND APPLICATIONS
_|_| |__| |  |    /   /        Edinburgh, 5-6th September, 1999


		. To be held in conjunction with VLDB int. conf.
		. Symposium URL: http://www.cs.rmit.edu.au/conf/doa99
		. Publisher: IEEE Press
              	. Sponsor: OMG, RMIT

SCOPE
-----

There is increasing agreement among IT researchers and practioners about
the importance, potential and advances of distributed object systems made in
recent years. These systems offer many promises for their use in various
applications, including telecommunications and more recently banking
applications. Distributed object systems offer solutions to technical
problems, including interoperability across different software and database
platforms. Distributed object systems are built according to different
paradigms and architectures, such as OMG's CORBA and other object request
broker principles and implementations, and contingent technologies such 
as SUN's Java and Microsoft's active objects, to provide a basis for
building complex distributed applications.
 
The future success of distributed object systems will be not only dependent
on how the basic requirements (to develop open, scalable and reliable
distributed and heterogeneous applications and platforms) are met but also
how the underlying distributed object technology can be integrated with
existing complementary technologies and applications, such as WWW, multimedia
and databases. Also legacy systems may sometimes substantially benefit from
a reengineering effort using distributed objects, e.g. to turn them into data
warehouses.
 
During this symposium we want attendees to be able to evaluate existing ORB
middleware products; to propose solutions to major limitations of existing
products; and to introduce promising future research directions for
distributed objects. We are particularly interested in the evaluation of
existing distributed object systems and how they are used to design and to
implement large scale industrial distributed applications. We are also
seeking theoretical and practical papers addressing innovating issues related
to distributed objects.

TOPICS OF INTEREST
------------------

The topics of this symposium include, but are not limited to:
	. Distributed business objects
	. Distributed and mobile agents
	. Design patterns for distributed object design
	. Database services, in particular, persistency, transaction, query 
	  and replication services
	. Intelligent traders
	. Interoperability supporting environments
	. Integration with database systems and interfaces
	. Methodologies to develop distributed object applications,
	  to reintegrate legacy systems, and to design CORBA-based applications
	. Multimedia distributed objects
	. Multicast protocols for distributed objects
	. Object caching
	. Reliability, fault-tolerance and recovery
	. Real-time ORB middleware
	. Security
	. Specification and enforcement of quality of service
	. Wrapper libraries and wrapper implementation strategies

In order to maximise the benefits of co-location with VLDB'98, 
authors are asked to give particular emphasis to issues relating to
the deployment of distributed object technology in the provision
and utilisation of database systems.

IMPORTANT DATES
---------------

	Electronic submission deadline:	April 1st, 1999
	Notification of acceptance:  	May 15th, 1999
	Camera-ready copies: 		June 15th, 1999
	Symposium: 			September 5/6th, 1999


SUBMISSION DETAILS
------------------

Submitted papers will be carefully evaluated based on originality, 
significance, technical soundness, and clarity of expression. All papers
will be refereed by at least three members of the program committee.
Papers should be in English and must not exceed 8000 words. Submissions
must be in the form of a single uuencoded compressed PostScript file sent
by e-mail to

		zahirt@cs.rmit.edu.au
 
accompanied by a separate email message with the following information
on the paper:

	. title
	. author(s)
	. affiliation(s)
	. e-mail and address of the contact author

Please make sure that your PostScript file can be previewed with GhostScript
and is printable on a standard PostScript printer. If electronic submission is
not possible, please contact

	Zahir Tari
	zahirt@cs.rmit.edu.au
 
to make special arrangements, at least one week before the deadline.
 
The proceedings will be published by IEEE Press.


ORGANISING COMMITTEE
--------------------

. GENERAL CHAIR
	Asuman Dogac
	Middle East Technical University
	Ankara, Turkey
	asuman@srdc.metu.edu.tr

. PROGRAM COMMITTEE CO-CHAIRS
	Omran Bukhres	(USA)
	Robert Meersman	(Belgium)
	Zahir Tari	(Australia)

. ORGANISING CHAIR
	Peter Thanisch
	The University of Edinburgh
	Department of Computer Science 
	Mayfield Road, Edinburgh EH9 3JZ 
	Scotland 
	(Fax) 0131 667 7209
	(E-mail) pt@dcs.ed.ac.uk

. PUBLICITY CHAIR
	Sam Makki
	RMIT Dept. of Computer Science
	sam@cs.rmit.edu.au

. PROGRAM COMMITTEE
	Gustavo Alonso (ETHZ, Zurich) 
	Bill Appelbe (RMIT, Australia) 
        Jose Blakeley (Microsoft, USA) 
        Anthony Bloesch (Visio, USA) 
        Sjaak Brinkkemper (Baan, The Netherlands) 
        David Curtis (OMG, USA) 
        Klaus Dittrich (University of Zurich, Switzerland) 
        Chris Gokey (NASA, USA) 
        Rachid Guerraoui (EPFL, Switzerland) 
        Slimane Hammoudi (Univ. of Minho, Portugal) 
        Dimitris Karagiannis (University of Vienna and B.O.C. GmbH, Austria) 
        Sacha Krakowiak (University of Grenoble, France) 
        Roger King (University of Collorado, USA) 
        Ling Liu (Oregon Graduate Institute, USA) 
        Frank Manola (Object Services and Consulting, USA) 
        Michele Missikoff (CNR Roma, Italy) 
        Tom Northcutt (NASA, USA) 
        Tamer Ozsu (University of Alberta, Cannada) 
        Mike P. Papazoglou (Tilburg University, The Netherlands) 
        Richard Soley (OMG, USA) 
        Kerry Raymond (DSTC, Australia) 
        Hakki Toroslu (Middle East Technical University, Turkey) 
        Irv Traiger (IBM Santa Teresa Lab, USA) 
        Arkady Zaslavsky (Monash University, Australia) 
        Roberto Zicari (Johann Wolfgang Goethe-Univ., Germany) 
        Wilfried Verachtert (MediaGenix, Belgium) 
        Andreas Vogel (Visigenic Software, USA) 
        Andrew Watson (OMG, USA)

SYMPOSIUM ATTENDANCE
--------------------

Authors of accepted papers will be expected to present their work at the
symposium. Attendees can also register to VLDB'99 conference which will be
held early. Details about the VLDB'99 conference can be found at
              http://www.cs.rmit.edu.au/conf/doa99







From rem-conf Mon Jan 18 20:11:13 1999 
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Date: Mon, 18 Jan 1999 23:04:51 -0500
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
Organization: Bell Laboratories
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To: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
CC: casner@cisco.com, rem-conf@es.net
Subject: Parity FEC draft and MIME and SDP
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In preparation of the parity FEC draft for last call, I ran into two
open issues for which we need resolution. Both surround signaling of the
use of parity within SDP. Since rfc2198 addresses SDP issues, it seems
appropriate for them to be addressed in the parity draft.

The first issue is that we need a new MIME type for the parity format.
We can either define FEC as a sub-type of application, since it is
neither audio or video specific:

application/xor

or define xor as a subtype of both audio and video:

audio/xor
video/xor

Other names besides xor could be fec or parity. I'd probably vote for
parity. 

With that done, the next issue is how to signal the usage of parity
within SDP.

Method 1
--------

Since FEC is a separate stream, it makes sense to define it as a new
media type:

m=fec 12345 RTP/AVP 110
m=audio 12345 RTP/AVP 0 1
a=rtpmap:110 xor/8000/1

This says that the FEC goes to the same port/address as the audio, and
its using payload type 110. Problem here is that the m lines are
supposed to be top level MIME types, and fec is not a top level MIME
type. So, we could alternatively do:

m=application 12345 RTP/AVP 110
m=audio 12345 RTP/AVP 0 1
a=rtpmap:110 parity/8000/1


Method 2
---------
This approach is more like rfc2198:

m=audio 12345 RTP/AVP 0 1 110
a=rtpmap:110 xor/8000/1

The problem with this is that the FEC can go on a separate multicast
group or port from the media itself, which can't be expressed in the
above fashion. This is different from rfc2198, where the media and FEC
are bound together. 

If one were using rfc2198 to carry the FEC as a redundant payload, then
something like:

m=audio 12345 RTP/AVP 121 0 110
a=rtpmap:121 red/8000/1
a=rtpmap:110 xor/8000/1
a=fmtp:121 0/110

seems right.

Also, is it useful to indicate the payload types likely to be protected
by parity? Something like:

a=fmtp:110 0

Would indicate that payload type 0 is protected by FEC (the FEC payload
type is 110). Is this needed?


So, my proposal would be::


* use application/parity as the MIME type for FEC
* use a separate media line, of the form m=application <port> RTP/AVP
<pti>, and an rtpmap of the form a=rtpmap:<pti> parity/8000/1, to
indicate FEC.
* allow an fmtp line to indicate the set of codecs covered by the FEC
* when rfc2198 is used to carry FEC, indicate the FEC as a payload type
for one of the redundant codecs, and then use an rtpmap to indicate that
the codec is FEC.

Comments?

-Jonathan R.
-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX: (732) 834-5379                         Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Mon Jan 18 23:01:41 1999 
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From rem-conf Tue Jan 19 01:18:10 1999 
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Date: Tue, 19 Jan 1999 01:08:38
To: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
From: Ross Finlayson <finlayson@live.com>
Subject: Re: Parity FEC draft and MIME and SDP
Cc: rem-conf@es.net
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Actually, I  think I prefer your "Method 1" (for the case where FEC is
carried as a separate stream):

>Since FEC is a separate stream, it makes sense to define it as a new
>media type:
>
>m=fec 12345 RTP/AVP 110
>m=audio 12345 RTP/AVP 0 1
>a=rtpmap:110 xor/8000/1
>
>This says that the FEC goes to the same port/address as the audio, and
>its using payload type 110.

>Problem here is that the m lines are supposed to be top level MIME types

I don't think we need to be so anal retentive about this, as m= lines are
not actually *used* as MIME types.  (Also, is "control" a top-level MIME
type?)

In this case "fec" clearly describes the medium - one which the
SDP-handling application (e.g., session launcher) may know about and choose
to handle (or ignore) independently of the audio/video tool.  In contrast,
"m=application ..." seems pretty pointless - it adds another level of
indirection for no particular good reason.

	Ross.





From rem-conf Tue Jan 19 03:33:23 1999 
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To: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
cc: casner@cisco.com, rem-conf@es.net
Subject: Re: Parity FEC draft and MIME and SDP
In-reply-to: Your message of "Mon, 18 Jan 1999 23:04:51 EST." <36A40463.A6288007@dnrc.bell-labs.com>
Date: Tue, 19 Jan 1999 11:30:35 +0100
Message-ID: <911.916745435@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> Jonathan Rosenberg writes:
>In preparation of the parity FEC draft for last call, I ran into two
>open issues for which we need resolution. Both surround signaling of the
>use of parity within SDP. Since rfc2198 addresses SDP issues, it seems
>appropriate for them to be addressed in the parity draft.
>
>The first issue is that we need a new MIME type for the parity format.
>We can either define FEC as a sub-type of application, since it is
>neither audio or video specific:
>
>application/xor
>
>or define xor as a subtype of both audio and video:
>
>audio/xor
>video/xor
>
>Other names besides xor could be fec or parity. I'd probably vote for
>parity. 

I'd prefer the FEC schemes to be registered under both audio and video,
seems to make things easier when defining the SDP.

How about audio/parityfec and video/parityfec?

>With that done, the next issue is how to signal the usage of parity
>within SDP.
>
>Method 1
>--------
>
>Since FEC is a separate stream, it makes sense to define it as a new
>media type:
>
>m=fec 12345 RTP/AVP 110
>m=audio 12345 RTP/AVP 0 1
>a=rtpmap:110 xor/8000/1
>
>This says that the FEC goes to the same port/address as the audio, and
>its using payload type 110. Problem here is that the m lines are
>supposed to be top level MIME types, and fec is not a top level MIME
>type. So, we could alternatively do:
>
>m=application 12345 RTP/AVP 110
>m=audio 12345 RTP/AVP 0 1
>a=rtpmap:110 parity/8000/1

Both of these seem kind-of ugly... what we have is audio protected by FEC,
so it makes sense to label it as audio -- that way the tool parsing the SDP
knows what sort of media tool to start.

>Method 2
>---------
>This approach is more like rfc2198:
>
>m=audio 12345 RTP/AVP 0 1 110
>a=rtpmap:110 xor/8000/1
>
>The problem with this is that the FEC can go on a separate multicast
>group or port from the media itself, which can't be expressed in the
>above fashion. This is different from rfc2198, where the media and FEC
>are bound together. 

If the FEC is sent as a separate stream, it should have a separate m= line.
The problem is linking the FEC stream to the media, but that can be solved
with "a=fmtp:" giving the media port to associate the FEC with...

	m=audio 1234 RTP/AVP 0 1 
	m=audio 1235 RTP/AVP 110
	a=rtpmap:110 xor/8000/1
	a=fmtp:110 1234

Not sure how that would work if the media was carried on a different
address to the FEC though. Is that something we need to worry about?

Colin



From rem-conf Tue Jan 19 07:16:06 1999 
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From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
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To: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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Subject: Re: Parity FEC draft and MIME and SDP
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Colin Perkins wrote:
> 
> >Method 1
> >--------
> >
> >Since FEC is a separate stream, it makes sense to define it as a new
> >media type:
> >
> >m=fec 12345 RTP/AVP 110
> >m=audio 12345 RTP/AVP 0 1
> >a=rtpmap:110 xor/8000/1
> >
> >This says that the FEC goes to the same port/address as the audio, and
> >its using payload type 110. Problem here is that the m lines are
> >supposed to be top level MIME types, and fec is not a top level MIME
> >type. So, we could alternatively do:
> >
> >m=application 12345 RTP/AVP 110
> >m=audio 12345 RTP/AVP 0 1
> >a=rtpmap:110 parity/8000/1
> 
> Both of these seem kind-of ugly... what we have is audio protected by FEC,
> so it makes sense to label it as audio -- that way the tool parsing the SDP
> knows what sort of media tool to start.

The first method would also allow you to know this since the a:fmtp line
would indicate which types are being protected by FEC. Its definitely
easier, though, if its specifically an m=audio or m=video line, though.

> 
> >Method 2
> >---------
> >This approach is more like rfc2198:
> >
> >m=audio 12345 RTP/AVP 0 1 110
> >a=rtpmap:110 xor/8000/1
> >
> >The problem with this is that the FEC can go on a separate multicast
> >group or port from the media itself, which can't be expressed in the
> >above fashion. This is different from rfc2198, where the media and FEC
> >are bound together.
> 
> If the FEC is sent as a separate stream, it should have a separate m= line.
> The problem is linking the FEC stream to the media, but that can be solved
> with "a=fmtp:" giving the media port to associate the FEC with...
> 
>         m=audio 1234 RTP/AVP 0 1
>         m=audio 1235 RTP/AVP 110
>         a=rtpmap:110 xor/8000/1
>         a=fmtp:110 1234
> 
> Not sure how that would work if the media was carried on a different
> address to the FEC though. Is that something we need to worry about?

Once its a separate m line, you can use a c line below the media line to
indicate a different address. Also, I don't see the need for identifying
the port number of the media in the a=fmtp line; why not indicate the
payload type being protected? So then:


m=audio 1234 RTP/AVP 0 1
c=IN IP4 224.2.1.1/127
m=audio 1235 RTP/AVP 110
c=IN IP4 224.2.1.2/127
a=rtpmap:110 parityfec/8000/1
a=fmtp:110 0

This indicates that the FEC is on a group one higher, and port one
higher, than the audio, and that the FEC is PTI 110, and it is
protecting the audio with payload type 0.

audio/parityfec and video/parityfec are fine by me; I agree these are
more intuitive than application/parityfec.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX:   (732) 834-5379                       Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Tue Jan 19 08:14:17 1999 
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From: Mark Handley <mjh@east.isi.edu>
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To: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
cc: Colin Perkins <C.Perkins@cs.ucl.ac.uk>, casner@cisco.com, rem-conf@es.net
Subject: Re: Parity FEC draft and MIME and SDP 
In-reply-to: Your message of "Tue, 19 Jan 1999 10:06:06 EST."
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>Once its a separate m line, you can use a c line below the media line to
>indicate a different address. Also, I don't see the need for identifying
>the port number of the media in the a=fmtp line; why not indicate the
>payload type being protected? So then:
>
>
>m=audio 1234 RTP/AVP 0 1
>c=IN IP4 224.2.1.1/127
>m=audio 1235 RTP/AVP 110
>c=IN IP4 224.2.1.2/127
>a=rtpmap:110 parityfec/8000/1
>a=fmtp:110 0
>
>This indicates that the FEC is on a group one higher, and port one
>higher, than the audio, and that the FEC is PTI 110, and it is
>protecting the audio with payload type 0.

This isn't sufficient - I can have more than one audio media in a
session, and even have different media with the same dynamic payload
type.  Eg, the following is valid:

m=audio 1234 RTP/AVP 0
c=IN IP4 224.2.1.1/127
a=lang:en
m=audio 1234 RTP/AVP 0
c=IN IP4 224.2.1.2/127
a=lang:fr
m=audio 1235 RTP/AVP 110
c=IN IP4 224.2.1.3/127
a=rtpmap:110 parityfec/8000/1
a=fmtp:110 pt=0;addr=224.2.1.1;port=1234

The two audio streams are alternatives, and in this case I've even put
them on the same port.  In this case you can only distinguish them by
address.

An alternative is to add a tag attribute:

m=audio 1234 RTP/AVP 0
c=IN IP4 224.2.1.1/127
a=lang:en
a=tag:1
m=audio 1234 RTP/AVP 0
c=IN IP4 224.2.1.2/127
a=lang:fr
a=tag:2
m=audio 1235 RTP/AVP 110
c=IN IP4 224.2.1.3/127
a=rtpmap:110 parityfec/8000/1
a=fmtp:110 tag=1

The tag attribute would be useful in general, but might be difficult
to implement in a backward compatible way because unlike most tags,
it's not optional to understand it.

Our problem with the mime type is because FEC isn't really a media
type, but more analogous to a content-transfer-encoding.

I'd be willing to argue for a top-level SDP type of "encoding" to
serve this function:

m=audio 1234 RTP/AVP 0
c=IN IP4 224.2.1.1/127
a=tag:1
m=encoding 1235 RTP/AVP 110
c=IN IP4 224.2.1.3/127
a=rtpmap:110 parityfec/8000/1
a=fmtp:110 tag=1

This would mean that tools wouldn't mistakenly believe this is a new
media stream.  

It doesn't quite fit with the
"SDP-media-type = mime-toplevel-content-type" rule in the SDP spec 
though, so we might have to issue a new SDP ID including this and stating
the exception.

Finally, another alternative is to put the FEC information entirely in
an fmtp attribute:

m=audio 1234 RTP/AVP 0 110
c=IN IP4 224.2.1.1/127
a=rtpmap:110 parityfec/8000/1
a=fmtp:110 addr=IN/IP4/224.2.1.2;port=1235

In this case there's no possibility of confusing it with being a new
medium intended for presentation to the user.  However, it's slightly
misleading because the FEC data doesn't go to 224.2.1.1/1234 but to
224.2.1.2/1235

Cheers,
	Mark



From rem-conf Tue Jan 19 08:28:53 1999 
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To: Mark Handley <mjh@east.isi.edu>
cc: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>, casner@cisco.com, 
    rem-conf@es.net
Subject: Re: Parity FEC draft and MIME and SDP
In-reply-to: Your message of "Tue, 19 Jan 1999 11:06:20 EST." <19853.916761980@north.lcs.mit.edu>
Date: Tue, 19 Jan 1999 16:20:25 +0100
Message-ID: <2008.916762825@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> Mark Handley writes:
...
>An alternative is to add a tag attribute:
>
>m=audio 1234 RTP/AVP 0
>c=IN IP4 224.2.1.1/127
>a=lang:en
>a=tag:1
>m=audio 1234 RTP/AVP 0
>c=IN IP4 224.2.1.2/127
>a=lang:fr
>a=tag:2
>m=audio 1235 RTP/AVP 110
>c=IN IP4 224.2.1.3/127
>a=rtpmap:110 parityfec/8000/1
>a=fmtp:110 tag=1
>
>The tag attribute would be useful in general, but might be difficult
>to implement in a backward compatible way because unlike most tags,
>it's not optional to understand it.

Or maybe...

m=audio 1234 RTP/AVP 0
c=IN IP4 224.2.1.1/127
a=lang:en
a=tag:1
m=audio 1234 RTP/AVP 0
c=IN IP4 224.2.1.2/127
a=lang:fr
a=tag:2
m=audio 1235 RTP/AVP 110
c=IN IP4 224.2.1.3/127
a=rtpmap:110 parityfec/8000/1
a=fmtp:110
a=depends-on-tag:1

which allows to indicate that the FEC is only useful if a particular media
stream is also being received. 

We could also do "a=alternative-to-tag:" to make that explicit, giving:

m=audio 1234 RTP/AVP 0
c=IN IP4 224.2.1.1/127
a=lang:en
a=tag:1
a=alternative-to-tag:2
m=audio 1234 RTP/AVP 0
c=IN IP4 224.2.1.2/127
a=lang:fr
a=tag:2
a=alternative-to-tag:1
m=audio 1235 RTP/AVP 110
c=IN IP4 224.2.1.3/127
a=rtpmap:110 parityfec/8000/1
a=fmtp:110
a=depends-on-tag:1

Okay, we need to think of shorter a= names, but is the idea sensible?

Colin



From rem-conf Tue Jan 19 08:51:23 1999 
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CC: Mark Handley <mjh@east.isi.edu>, casner@cisco.com, rem-conf@es.net
Subject: Re: Parity FEC draft and MIME and SDP
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Colin Perkins wrote:
> 
> --> Mark Handley writes:
> ...
> >An alternative is to add a tag attribute:
> >
> >m=audio 1234 RTP/AVP 0
> >c=IN IP4 224.2.1.1/127
> >a=lang:en
> >a=tag:1
> >m=audio 1234 RTP/AVP 0
> >c=IN IP4 224.2.1.2/127
> >a=lang:fr
> >a=tag:2
> >m=audio 1235 RTP/AVP 110
> >c=IN IP4 224.2.1.3/127
> >a=rtpmap:110 parityfec/8000/1
> >a=fmtp:110 tag=1
> >
> >The tag attribute would be useful in general, but might be difficult
> >to implement in a backward compatible way because unlike most tags,
> >it's not optional to understand it.
> 
> Or maybe...
> 
> m=audio 1234 RTP/AVP 0
> c=IN IP4 224.2.1.1/127
> a=lang:en
> a=tag:1
> m=audio 1234 RTP/AVP 0
> c=IN IP4 224.2.1.2/127
> a=lang:fr
> a=tag:2
> m=audio 1235 RTP/AVP 110
> c=IN IP4 224.2.1.3/127
> a=rtpmap:110 parityfec/8000/1
> a=fmtp:110
> a=depends-on-tag:1
> 
> which allows to indicate that the FEC is only useful if a particular media
> stream is also being received.
> 
> We could also do "a=alternative-to-tag:" to make that explicit, giving:
> 
> m=audio 1234 RTP/AVP 0
> c=IN IP4 224.2.1.1/127
> a=lang:en
> a=tag:1
> a=alternative-to-tag:2
> m=audio 1234 RTP/AVP 0
> c=IN IP4 224.2.1.2/127
> a=lang:fr
> a=tag:2
> a=alternative-to-tag:1
> m=audio 1235 RTP/AVP 110
> c=IN IP4 224.2.1.3/127
> a=rtpmap:110 parityfec/8000/1
> a=fmtp:110
> a=depends-on-tag:1
> 
> Okay, we need to think of shorter a= names, but is the idea sensible?

I'm not clear on the purpose of the alternative-to-tag attribute. Are
you saying that its purpose is to indicate that the first and second
media streams are alternatives?

Another way to associate the fec stream to the media stream is
implicitly by positioning; if a media stream thats FEC exists, it is
associated with the m line above it. So:

m=audio 1234 RTP/AVP 0
c=IN IP4 224.2.1.1/127
m=audio 1234 RTP/AVP 110
a=rtpmap:110 parityfec/8000/1
c=IN IP4 224.1.2.3/127
m=audio 1235 RTP/AVP 0
c=IN IP4 224.2.1.1/127
m=audio 1235 RTP/AVP 110
a=rtpmap:110 parityfec/8000/1
c=IN IP4 224.1.2.3/127

There is then no ambiguity. The first audio FEC stream is associated
with the media on port 1234, and the second for the media on port 1235.
This avoids the messy use of tags.

I think this is still backwards compatible. A tool that doesn't
understand FEC might think there are four media streams, but anyway
won't be able to parse the two FEC ones since they use a codec that is
not understood (parityfec). In fact, same should hold true for the tag
solutions.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX:   (732) 834-5379                       Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Tue Jan 19 08:51:23 1999 
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From: Mark Handley <mjh@east.isi.edu>
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X-Phone: +1 617 253 6011
To: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
cc: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>, casner@cisco.com,
        rem-conf@es.net
Subject: Re: Parity FEC draft and MIME and SDP 
In-reply-to: Your message of "Tue, 19 Jan 1999 16:20:25 +0100."
             <2008.916762825@cs.ucl.ac.uk> 
Date: Tue, 19 Jan 1999 11:47:32 -0500
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>Or maybe...
>
>m=audio 1234 RTP/AVP 0
>c=IN IP4 224.2.1.1/127
>a=lang:en
>a=tag:1
>m=audio 1234 RTP/AVP 0
>c=IN IP4 224.2.1.2/127
>a=lang:fr
>a=tag:2
>m=audio 1235 RTP/AVP 110
>c=IN IP4 224.2.1.3/127
>a=rtpmap:110 parityfec/8000/1
>a=fmtp:110
>a=depends-on-tag:1
>
>which allows to indicate that the FEC is only useful if a particular media
>stream is also being received. 

Something we've been bad at in the past, but we need to be careful to
find the right split between what we expect from the sdp-tool and what
we expect from the media tool.

In this case, I read your use of the depends-on-tag attribute to mean
that there's a constraint on receiving the medium that we must also
receive this other medium.  Fine so far.  But this is a general SDP
thing, and is not FEC specific.  What you want is *additionally* for
the FEC decoder to have access to the information about the stream it
depends on.  If you want to have the depends-on-tag attribute to have
generic meaning, you probably don't want to overload it's meaning in
the context of FEC.

I'd prefer something that specifies both relationships separately, and
also avoids using the audio media type.

m=audio 1234 RTP/AVP 0
c=IN IP4 224.2.1.1/127
a=lang:en
a=tag:1
m=audio 1234 RTP/AVP 0
c=IN IP4 224.2.1.2/127
a=lang:fr
a=tag:2
m=encoding 1235 RTP/AVP 110
c=IN IP4 224.2.1.3/127
a=rtpmap:110 parityfec/8000/1
a=fmtp:110 tag=1
a=req:1


This is rather getting off the original topic, but:

>We could also do "a=alternative-to-tag:" to make that explicit,
>giving:

An alternative way to phrase the same thing is to have a attribute
that specifies a group-of-alternatives.  Then you don't need the tag
field unless it's required by something else.

m=audio 1234 RTP/AVP 0
c=IN IP4 224.2.1.1/127
a=lang:en
a=goa:1
m=audio 1234 RTP/AVP 0
c=IN IP4 224.2.1.2/127
a=lang:fr
a=goa:1

Anything with the same goa tag is one of a set of alternative streams.
You realise this is a slippery slope...

Cheers,
	Mark



From rem-conf Tue Jan 19 09:00:34 1999 
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cc: Colin Perkins <C.Perkins@cs.ucl.ac.uk>, casner@cisco.com, rem-conf@es.net
Subject: Re: Parity FEC draft and MIME and SDP 
In-reply-to: Your message of "Tue, 19 Jan 1999 11:43:55 EST."
             <36A4B64B.66E687B@dnrc.bell-labs.com> 
Date: Tue, 19 Jan 1999 11:58:54 -0500
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>Another way to associate the fec stream to the media stream is
>implicitly by positioning; if a media stream thats FEC exists, it is
>associated with the m line above it. So:
>
>m=audio 1234 RTP/AVP 0
>c=IN IP4 224.2.1.1/127
>m=audio 1234 RTP/AVP 110
>a=rtpmap:110 parityfec/8000/1
>c=IN IP4 224.1.2.3/127
>m=audio 1235 RTP/AVP 0
>c=IN IP4 224.2.1.1/127
>m=audio 1235 RTP/AVP 110
>a=rtpmap:110 parityfec/8000/1
>c=IN IP4 224.1.2.3/127

Unfortunately the SDP semantics give no meaning to the ordering of the
media, and in this case there's no way for the SDP parser to know that
the second medium is not a medium to be presented to the user at all.

With hindsight, we made a number of mistakes with SDP, and one of them
is that too much of the meaning is implicit.  If we need to add
semantics, I would like to see the semantics as explicit as possible
given the limitations of the syntax.

Cheers,
	Mark



From rem-conf Tue Jan 19 09:31:33 1999 
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  Seminar
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Berkeley Multimedia, Interfaces and Graphics Seminar

Simplifying the Controls of an Interactive Movie Game 

(Wednesday January 20, 1999 12:30-2:00 PDT 405 Soda Hall) 

Jeff Johnson 
UI Wizards, Inc. 

Eight months before an interactive movie game was due to be shipped, its
developers and funders decided that its user interface had to be radically
simplified. The author was given the task of designing a new, simpler
control scheme. This paper describes the redesign process, the design
issues that arose and how they were resolved, the tests that were conducted
to evaluate new design ideas, and concludes with an evaluation of the
resulting design, lessons learned, and thoughts on user-interface design
vs. game design. 

Reference: 
Johnson, J., Simplifying the Controls of an Interactive Movie Game,
Proceedings of ACM CHI'98, pages 65-72. 

The seminar is broadcast on the Internet Mbone. The addresses are video
224.2.163.7/57076 and audio 224.2.147.61/27202. 

Sponsored by the Berkeley Multimedia Research Center

____________________________________________

Florissa Colina
Berkeley Multimedia Research Center (BMRC)
http://bmrc.berkeley.edu/
626 Soda Hall #1776
University of California
Berkeley, CA 94720-1776
Phone: (510) 643-0800   FAX: (510) 642-5615  
Email: florissac@bmrc.berkeley.edu



From rem-conf Tue Jan 19 17:36:50 1999 
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From: "Ram Kordale" <kordale@relic.engr.sgi.com>
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I could not find any relevant discussion in the confctrl and rem-conf archives.
So, I am posting this here...

Am I right in stating that an RTSP session is kept alive at a server until one
of the following happens:
  (1) A TEARDOWN is received for that session
  (2) There is no activity on the RTSP channel for a period specified in the
      timeout parameter in the Session response header (or a default of 60
            seconds)

I fail to see the usefulness of (2) given that you can expect long periods of
inactivity on the control channel in applications using RTSP (for example, when
users watch movies).

Moreover, many of the "dumb boxes" (like many settop boxes) do not send
TEARDOWN messages when users switch them off.

One way to solve this problem would be to have clients send periodic keep-alive
messages.

Any comments?

-- 
Ram Kordale
Advanced Media Products



From rem-conf Tue Jan 19 18:53:17 1999 
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Date: Tue, 19 Jan 1999 21:45:14 -0500
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
Organization: Bell Laboratories
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To: Mark Handley <mjh@east.isi.edu>
CC: Colin Perkins <C.Perkins@cs.ucl.ac.uk>, casner@cisco.com, rem-conf@es.net
Subject: Re: Parity FEC draft and MIME and SDP
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Mark Handley wrote:
> 
> >which allows to indicate that the FEC is only useful if a particular media
> >stream is also being received.
> 
> Something we've been bad at in the past, but we need to be careful to
> find the right split between what we expect from the sdp-tool and what
> we expect from the media tool.
> 
> In this case, I read your use of the depends-on-tag attribute to mean
> that there's a constraint on receiving the medium that we must also
> receive this other medium.  Fine so far.  But this is a general SDP
> thing, and is not FEC specific.  What you want is *additionally* for
> the FEC decoder to have access to the information about the stream it
> depends on.  If you want to have the depends-on-tag attribute to have
> generic meaning, you probably don't want to overload it's meaning in
> the context of FEC.
> 
> I'd prefer something that specifies both relationships separately, and
> also avoids using the audio media type.
> 
> m=audio 1234 RTP/AVP 0
> c=IN IP4 224.2.1.1/127
> a=lang:en
> a=tag:1
> m=audio 1234 RTP/AVP 0
> c=IN IP4 224.2.1.2/127
> a=lang:fr
> a=tag:2
> m=encoding 1235 RTP/AVP 110
> c=IN IP4 224.2.1.3/127
> a=rtpmap:110 parityfec/8000/1
> a=fmtp:110 tag=1
> a=req:1
> 

If indeed you believe there are two separate things here - (1) that
stream a depends on stream b, and (2) the FEC is over stream b, I'd
rather stick to solving just the problem at hand instead of doing the
other more general extension. 

So, how about:

* using the encoding m line to indicate FEC
* add a tag attribute for other m lines
* define the fmtp line to indicate which media stream (via the tag) is
being protected

If we do more general extensions later (group of alternatives) we need
the tags anyway and they can be used for that too.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX: (732) 834-5379                         Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Wed Jan 20 13:23:34 1999 
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From: "Ethendranath Bommaiah" <ethenb@hotmail.com>
To: kordale@relic.engr.sgi.com, rem-conf@es.net
Cc: ethen@dnrc.bell-labs.com
Subject: Re: RTSP keep-alives?
Date: Wed, 20 Jan 1999 13:11:50 PST
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Current implementation of RTSP in Real G2 player/server uses ping
messages from the server to the client. The ping message looks
something like:
SET_PARAMETER * RTSP/1.0
CSeq: 1
Ping: Pong

the response to which looks like:
RTSP/1.0 451 Parameter Not Understood
CSeq: 1
Date: Sat, 12 Dec 1998 18:51:08 GMT

Server is taking the initiative here which is probably okay -- what
if the client is down ? Also, in case of playing long movies, the server 
should probably set the timer for timeout only after playing out the 
whole movie. If RTCP receiver reports are not received from
the receiver, then maybe you should stop playing out the movie (on
RTP) and start RTSP timer ? In some sense RTCP receiver reports serve
as keep-alive messages for both RTSP and RTP. If RTCP is not supported, 
then we need RTSP keep alive messages -- which I guess
should be server initiated.

Ethen

>I could not find any relevant discussion in the confctrl and rem-conf 
archives.
>So, I am posting this here...
>
>Am I right in stating that an RTSP session is kept alive at a server 
until one
>of the following happens:
>  (1) A TEARDOWN is received for that session
>  (2) There is no activity on the RTSP channel for a period specified 
in the
>      timeout parameter in the Session response header (or a default of 
60
>            seconds)
>
>I fail to see the usefulness of (2) given that you can expect long 
periods of
>inactivity on the control channel in applications using RTSP (for 
example, when
>users watch movies).
>
>Moreover, many of the "dumb boxes" (like many settop boxes) do not send
>TEARDOWN messages when users switch them off.
>
>One way to solve this problem would be to have clients send periodic 
keep-alive
>messages.
>
>Any comments?
>
>-- 
>Ram Kordale
>Advanced Media Products
>
>


______________________________________________________
Get Your Private, Free Email at http://www.hotmail.com



From rem-conf Wed Jan 20 15:17:53 1999 
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From: "Ram Kordale" <kordale@relic.engr.sgi.com>
Message-Id: <9901201514.ZM8740@relic.engr.sgi.com>
Date: Wed, 20 Jan 1999 15:14:41 -0800
In-Reply-To: "Ethendranath Bommaiah" <ethenb@hotmail.com>
        "Re: RTSP keep-alives?" (Jan 20,  1:11pm)
References: <19990120211151.2451.qmail@hotmail.com>
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To: "Ethendranath Bommaiah" <ethenb@hotmail.com>, rem-conf@es.net
Subject: Re: RTSP keep-alives?
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Yes. We can overload existing calls. The spec suggests using GET_PARAMETER with
no entity body for this purpose as well. I was actually asking why keep-alives
are not explicit as in RFC 2068.

Using RTCP reports for keep-alives is not exactly a good idea for two reasons:
1) RTP data transport cannot be assumed.
2) Having the RTCP channel close the RTSP session initiated by the RTSP channel
does not exactly make for good software either.

Thanks for your input.
Ram

On Jan 20,  1:11pm, Ethendranath Bommaiah wrote:
> Subject: Re: RTSP keep-alives?
>
> Current implementation of RTSP in Real G2 player/server uses ping
> messages from the server to the client. The ping message looks
> something like:
> SET_PARAMETER * RTSP/1.0
> CSeq: 1
> Ping: Pong
>
> the response to which looks like:
> RTSP/1.0 451 Parameter Not Understood
> CSeq: 1
> Date: Sat, 12 Dec 1998 18:51:08 GMT
>
> Server is taking the initiative here which is probably okay -- what
> if the client is down ? Also, in case of playing long movies, the server
> should probably set the timer for timeout only after playing out the
> whole movie. If RTCP receiver reports are not received from
> the receiver, then maybe you should stop playing out the movie (on
> RTP) and start RTSP timer ? In some sense RTCP receiver reports serve
> as keep-alive messages for both RTSP and RTP. If RTCP is not supported,
> then we need RTSP keep alive messages -- which I guess
> should be server initiated.
>
> Ethen
>

-- 
Ram Kordale
Advanced Media Products



From rem-conf Wed Jan 20 18:03:26 1999 
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From: "Christian J. van den Branden Lambrecht" <vdb@hpl.hp.com>
To: "Christian J. van den Branden Lambrecht" <vdb@hpl.hp.com>
Subject: Session on Internet Telephony at the '99 SPIE Intl Symposium on Voice, Video, and Data Communications
Date: Wed, 20 Jan 1999 17:59:29 -0800
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SPIE's International Symposium on
Voice, Video, and Data Communications

Conference on Multimedia Systems and Applications II (vv06)

Session on Standards and Systems for Internet Telephony

Dear Colleagues,

We are organizing a session to be held within the conference on Multimedia
Systems and Applications II of the SPIE's International Symposium on Voice,
Video, and Data Communications. The session is intended to gather topical
papers discussing Internet Telephony systems and standards. The session is
meant to be tutorial and discuss all the aspects of internet telephony
systems, including but not limited to:
- H.323 protocol stack,
- IETF standards for internet telephony,
- Network management for internet telephony,
- Signaling,
- Audio compression,
- Gateways/Gatekeepers,
- Quality of Service management,
- Interoperability.

We are seeking to bring together researchers and practitioners and we are
looking for review or tutorial papers as well as contributions describing
original work. We are now soliciting contributions for this particular
session. The timetable is as follows:

Abstract due date:				February 22, 1999
Abstract due date for on-site proceedings:		February 8, 1999
Manuscript due date:				August 23, 1999
Manuscript due date for on-site proceedings:		June 28, 1999

The conference will be held in Boston, MA, September 19-22, 1999 at the
Hynes Convention Center.

Further information, call for papers, etc. can be found at
http://www.spie.org/info/vv/. Further information for this session can be
obtained by email to vdb@hpl.hp.com.

Please feel free to forward this announcement to any interested person.


=============================================================
Christian J. van den Branden Lambrecht  Voice: 650-857-7658
Hewlett-Packard Laboratories              Fax: 650-857-4691
1501 Page Mill Road, MS 1U20            Email: vdb@hpl.hp.com
Palo Alto, CA 94304                            vdb@ieee.org
=============================================================



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From rem-conf Thu Jan 21 07:16:00 1999 
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Date: Thu, 21 Jan 1999 10:05:12 -0500 (EST)
From: Ethendranath Bommaiah <ethen@dnrc.bell-labs.com>
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To: Ram Kordale <kordale@relic.engr.sgi.com>
cc: Ethendranath Bommaiah <ethenb@hotmail.com>, rem-conf@es.net
Subject: Re: RTSP keep-alives?
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On Wed, 20 Jan 1999, Ram Kordale wrote:

> Using RTCP reports for keep-alives is not exactly a good idea for two reasons:
> 1) RTP data transport cannot be assumed.
> 2) Having the RTCP channel close the RTSP session initiated by the RTSP channel
> does not exactly make for good software either.

I agree with you that failures should be detected by RTSP independent of
data channel. But they are not totally independent: If the client suffers 
buffer underflow (detected on data channel) because of net congestion or
server failure, client might want to sent an RTSP message to the server...
this might happen much before keep-alive timer at client expires.

On Jan 20,  1:11pm, Ethendranath Bommaiah wrote:

> > Server is taking the initiative here which is probably okay -- what
> > if the client is down ? 					   ^^^^
    ^^^^^^^^^^^^^^^^^^^^^^^
Ooops ! I guess RTSP keep-alive timers should be maintained at both client
and the server...and when they expire they could send ping messages to the
other party before tearing the connection down ?? (to handle network
failures)

Thanks,
Ethen




From rem-conf Thu Jan 21 10:24:13 1999 
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From: "Ram Kordale" <kordale@relic.engr.sgi.com>
Message-Id: <9901211019.ZM9719@relic.engr.sgi.com>
Date: Thu, 21 Jan 1999 10:19:35 -0800
In-Reply-To: "Ethendranath Bommaiah" <ethenb@hotmail.com>
        "Re: RTSP keep-alives?" (Jan 20,  1:11pm)
References: <19990120211151.2451.qmail@hotmail.com>
X-Mailer: Z-Mail (3.2.3 08feb96 MediaMail)
To: "Ethendranath Bommaiah" <ethenb@hotmail.com>, rem-conf@es.net
Subject: Re: RTSP keep-alives?
Cc: ethen@dnrc.bell-labs.com
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On Jan 20,  1:11pm, Ethendranath Bommaiah wrote:
> Subject: Re: RTSP keep-alives?
>
> Current implementation of RTSP in Real G2 player/server uses ping
> messages from the server to the client. The ping message looks
> something like:
> SET_PARAMETER * RTSP/1.0
> CSeq: 1
> Ping: Pong
>
> the response to which looks like:
> RTSP/1.0 451 Parameter Not Understood
> CSeq: 1
> Date: Sat, 12 Dec 1998 18:51:08 GMT
>
> Server is taking the initiative here which is probably okay -- what

According to the spec, this will not necessarily work since the client is not
required to have a persistent transport connection for the length of the RTSP
session.


> > Using RTCP reports for keep-alives is not exactly a good idea for two
reasons:
> > 1) RTP data transport cannot be assumed.
> > 2) Having the RTCP channel close the RTSP session initiated by the RTSP
channel
> > does not exactly make for good software either.
>
> I agree with you that failures should be detected by RTSP independent of
> data channel. But they are not totally independent: If the client suffers
> buffer underflow (detected on data channel) because of net congestion or
> server failure, client might want to sent an RTSP message to the server...
> this might happen much before keep-alive timer at client expires.
>

Yes. This is a reasonable argument albeit only for the client side.

Periodic "RTSP keep-alives" from the client based on a "keep-alive" header in
the server's response to SETUP seems to be a clean solution in the general
case(regardless of data transport protocol etc.).

Ram

-- 
Ram Kordale
Advanced Media Products



From rem-conf Fri Jan 22 20:38:59 1999 
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From: dawnc@TOGETHER.NET
Subject: complete solution to your VIDEO DUPLICATION needs
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We at MIRRORCOM DUPLICATION would like to thank you for opening and reading
this E-mail. The purpose of it is to introduce our self's to you or your
company as the complete solution to your VIDEO DUPLICATION needs. Our goal
is to provide our clients with the best rates possible for VHS dubs, Custom
package design, and fulfillment of product. Please visit our web site at:
http://www.together.net/~dawnc for more information about us.
Or contact us at: 819-868-0428 .

Once again thank you for your time.





From rem-conf Sat Jan 23 08:37:26 1999 
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We at MIRRORCOM DUPLICATION would like to thank you for opening and readi=
ng this E-mail. The purpose of it is to introduce our self=D5s to you or =
your company as the complete solution to your VIDEO DUPLICATION needs. Ou=
r goal is to provide our clients with the best rates possible for VHS dub=
s, Custom package design, and fulfillment of product. Please visit our we=
b site at: www.together.net/~dawnc for more information about us.
Or contact us at: 819-868-0428 .

Once again thank you for your time.



From rem-conf Sun Jan 24 04:33:18 1999 
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From: "Anders Klemets (Exchange)" <anderskl@exchange.microsoft.com>
To: 'Ram Kordale' <kordale@relic.engr.sgi.com>, rem-conf@es.net
Subject: RE: RTSP keep-alives?
Date: Sun, 24 Jan 1999 04:20:35 -0800
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> Periodic "RTSP keep-alives" from the client based on a 
> "keep-alive" header in
> the server's response to SETUP seems to be a clean solution 
> in the general
> case(regardless of data transport protocol etc.).

This is indeed what the spec already prescribes.  The "Session" header in
the SETUP response may have an attribute, called "timeout".  It specifies
the timeout of the RTSP Session.  The client may send a dummy GET_PARAMETER
command, specifying that particular Session ID, in order to keep the Session
alive.

I don't think the spec says that the client does not need to send these kind
of "keep-alive" commands while data is streamed from the server.  Because
that would make assumptions about the data transport protocol.  If the
server is transmitting data over UDP, using unicast, the server may be able
to detect that the client has failed through the receipt of ICMP error
messages.  But that may not be true for multicast, or other transport or
network protocols.

Anders



From rem-conf Mon Jan 25 21:51:02 1999 
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Date: 25 Jan 99 9:05:43 PM
From: pomnbvwq@yahoo.com
Message-Id: <BcY6mEsJbyNS6>
Subject: Maximize your website's traffic. Increase your search engine rank!
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From rem-conf Wed Jan 27 01:34:14 1999 
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Date: Wed, 27 Jan 1999 11:20:36 +0200
To: rem-conf@es.net
From: Ronen Goory <roneng@radmail.rad.co.il>
Subject: RTP compression over Frame Relay
Cc: ezra_k@es.net
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Hi,

In draft-ietf-avt-crtp-05.txt section 5 - Negotiating Compression it is
written "The use of IP/UDP/RTP compression over a particular link is a
function of the link-layer protocol.  It is expected that such negotiation
will be defined separately for PPP [4], for example..."

I would like to know if some know how it can be implemented over HDLC or
Frame Relay lines, without a PPP encapsulation?

Any help is appreciated.

Thanks

Ronen Goory
RAD Data Communications
31 Habarzel st.
Tel-Aviv, Israel
Tel 972-3-6455433
Fax 972-3-6455305




From rem-conf Wed Jan 27 01:49:55 1999 
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From: Carsten Bormann <cabo@informatik.uni-bremen.de>
To: Ronen Goory <roneng@radmail.rad.co.il>
Cc: rem-conf@es.net, ezra_k@es.net
Subject: Re: RTP compression over Frame Relay
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Ronen Goory writes:
> Hi,
> 
> In draft-ietf-avt-crtp-05.txt section 5 - Negotiating Compression it is
> written "The use of IP/UDP/RTP compression over a particular link is a
> function of the link-layer protocol.  It is expected that such negotiation
> will be defined separately for PPP [4], for example..."
> 
> I would like to know if some know how it can be implemented over HDLC or
> Frame Relay lines, without a PPP encapsulation?

You can always configure the ends to agree on the parameters needed.
For an overview of these parameters, you might want to consult the IP
compression over PPP draft, draft-engan-ip-compress-02.txt.

Gruesse, Carsten



From rem-conf Wed Jan 27 10:36:36 1999 
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Date: Wed, 27 Jan 1999 13:33:16 -0500
From: Mathias Kretschmer <mathias@research.att.com>
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Hi guys,

I'm wondering if there is a payload type/profile defined for AAC since it is
not compatible with MPEG Audio Layer 1...3 (which is covered by MPA in RFC
1890).

tx,

Mathias

--
Mathias Kretschmer, AT&T Labs Research               
mathias@research.att.com
                                                          fax
++1-973-360-8455



From rem-conf Wed Jan 27 15:14:59 1999 
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Date: Wed, 27 Jan 1999 13:00:53 -0800
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Organization: VM Labs
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Hi Guys:

I am wondering is there any RFC that specifies "Traffic Shaping" in 
gateway(s) particularly for voice and video traffic.  I am thinking
more or less an "intelligent network" that change the rate(s) and
transcode various protocols in audio and video in either gateway(s)
or router(s), with some regards to the End-to-End QoS client/server
negotiation.  There is a big problem in a very heterogenous network
where all sort of clients (in terms of MIPS) co-exist.

regards, Wilson

Wilson C. Chung, VM Labs
wchung@vmlabs.com



From rem-conf Wed Jan 27 16:59:11 1999 
From rem-conf-request@es.net Wed Jan 27 16:59:11 1999
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From: "Prof. Alexandros Eleftheriadis" <eleft@ee.columbia.edu>
To: "Wilson C. Chung" <wchung@vmlabs.com>, <rem-conf@es.net>
Subject: RE: Traffic Shaping
Date: Wed, 27 Jan 1999 19:50:07 -0500
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> From: Wilson C. Chung [mailto:wchung@vmlabs.com]
> Sent: Wednesday, January 27, 1999 4:01 PM
> To: rem-conf@es.net
> Subject: Traffic Shaping
>
>
> Hi Guys:
>
> I am wondering is there any RFC that specifies "Traffic Shaping" in
> gateway(s) particularly for voice and video traffic.  I am thinking
> more or less an "intelligent network" that change the rate(s) and
> transcode various protocols in audio and video in either gateway(s)
> or router(s), with some regards to the End-to-End QoS client/server
> negotiation.  There is a big problem in a very heterogenous network
> where all sort of clients (in terms of MIPS) co-exist.
>
> regards, Wilson
>
> Wilson C. Chung, VM Labs
> wchung@vmlabs.com

Howdy. What you refer to is called rate shaping; traffic shaping implies
that the average rate before and after remains the same. With rate shaping
the output bit rate is reduced, keeping the same encoding format. I
introduced rate shaping some years ago in my thesis where I designed
algorithms for MPEG-1/2. A student of mine later on applied it on IP (using
TCP-based flow control to ensure fair sharing with HTTP/FTP/ETC), and also
did a real-time implementation for MPEG-1/MPEG-2 streams. You can find more
info in http://www.ee.columbia.edu/~eleft and
http://www.ee.columbia.edu/~sej. Other techniques have been proposed since,
playing with different parts of the bitstream than I did, and making varied
claims in terms of performance/complexity. However, I don't know of any
implementation of them (let me know if you want paper refs).

For transcoding, I recall a couple of messages flying around this list a
year or so ago, referring to transcoding software, but I don't have
immediately accessible info. Transcoding makes most sence if there is really
a substantial difference between the source and target rates. It can also be
quite expensive. It's much easier to fully utilize the features of your
original bitstream to scale things down -- and all specs have many.

Hope this helps.




From rem-conf Wed Jan 27 21:28:03 1999 
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Date: Wed, 27 Jan 1999 21:20:22
To: mbone@isi.edu, rem-conf@es.net
From: Ross Finlayson <finlayson@live.com>
Subject: A new tool for multicast MP3 audio streaming
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With more than 90% of the Internet's users still using low-bandwidth
connections, we do multicast a disservice if we continue to promote and use
it for high-bandwidth users only.

As part of my ongoing campaign to help rectify this, I've released a new
tool that can multicast "MP3" (i.e., MPEG 1 or 2, layer III) audio files -
either at their 'native' bitrate (usually 128 kbps), or transcoded at a
lower bitrate.

The tool - "liveCaster" - is available (free) from
	<http://www.live.com/liveCaster/>

Versions are currently available for Windows (95,98,or NT), Linux, FreeBSD,
Solaris/SPARC and SGI Irix.

To use liveCaster, just point it at a directory containing "*.mp3" files,
and follow the setup instructions.

Unlike "Shoutcast" (another MP3 streaming tool (unicast) that's been
getting a lot of attention lately), "liveCaster" currently transcodes MP3
to lower bandwidths not by decoding the MPEG stream then re-encoding it,
but instead by picking out (a subset of) the already-encoded samples from
the original data, then repackaging them into new, smaller MPEG frames.
This means, unfortunately, that (i) the audio quality is not as good as
decoding/re-encoding, and (ii) MPEG 1 files can't be transcoded to less
than 32 kbps.  (On the other hand, liveCaster is *much* less CPU-hungry,
and, because it doesn't require the presence of an MPEG encoder, runs on
many more platforms beyond Windows.)

Receiving multicast MP3 streams
===============================
Unfortunately there currently seem to be few RTP-capable MP3 players (i.e.,
supporting RTP payload type 14, using the format described in RFC 2250).
I'm currently compiling (on my web site) a list of such tools; if you
provide such a tool, please let me know.  (I already know about Roland
Parviainen's "mIR" tool, although I have yet to try it.)

A PLEA: Has anyone added MP3 decoding to "rat"?  If not, this would make a
great 'class project'.  (FYI, there is 'open source' MP3 decoding software
available at <http://dorifer.heim3.tu-clausthal.de/~olli/mpg123/>; it may
be possible to use this in "rat".)

Limitations of the current version of liveCaster:
- It doesn't do RTCP.  (Yes, I know, "naughty, naughty"; I'll fix this soon)
- I doesn't automatically create SDP announcements
    (instead, you'll need to announce your sessions 'by hand')
- It doesn't yet use MALLOC for multicast address allocation (of course)
- There's currently a bug in the transcoder - a small percentage of MP3 files
    seem to confuse it, causing it to generate garbage

One final note: Please try to respect intellectual property rights, and try
not to be cavalier about transmitting copyrighted MP3 content.  The record
companies are already freaking out over MP3 in general, and I'd hate to see
the deployment and use of multicast suffer because of 'piracy' fears.

	Ross.





From rem-conf Thu Jan 28 07:08:31 1999 
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To: Ross Finlayson <finlayson@live.com>
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Reply-To: rat-trap.cs.ucl.ac.uk@cs.ucl.ac.uk
Subject: Re: A new tool for multicast MP3 audio streaming
In-reply-to: Your message of "Wed, 27 Jan 1999 21:20:22." <3.0.5.16.19990127212022.2f6736fc@shell7.ba.best.com>
Date: Thu, 28 Jan 1999 14:55:28 +0100
Message-ID: <1182.917535328@cs.ucl.ac.uk>
From: Orion Hodson <O.Hodson@cs.ucl.ac.uk>
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<3.0.5.16.19990127212022.2f6736fc@shell7.ba.best.com>Ross Finlayson writes:
> A PLEA: Has anyone added MP3 decoding to "rat"?  If not, this would make a
> great 'class project'.  (FYI, there is 'open source' MP3 decoding software
> available at <http://dorifer.heim3.tu-clausthal.de/~olli/mpg123/>; it may
> bepossible to use this in "rat".)

Mark Handley suggested a while back that we add the mpg123 code (or
something similar) and it is just one of those things I have never got
around to doing.  If someone is interested it should be reasonably
straightforward to incorporate mpg123 decoder.  The sources for the
new codec interfaces in RAT are not yet available publicly, but if you
are interested send a note to rat-trap@cs.ucl.ac.uk and we'll forward
the codec components and some sample code.

If someone takes this on I'll add the hooks for RAT to read coded
files for transmission so it can send and receive layer-III files.

There also are hooks in the experimental version of RAT to the Windows
ACM codecs specifically for exploiting the Fraunhoffer Layer-3 codec
(and G.723.1 but there's a key needed for this).  The code is
currently suffering from bit-rot, but should get fixed for the next
release.

cheers
- Orion

=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
Networked Multimedia Research Group      Tel: ++(0)171 419 3704
Department of Computer Science           Fax: ++(0)171 419 1397
University College London, Gower Street
London, WC1E 6BT, UK.     
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-




From rem-conf Thu Jan 28 07:16:22 1999 
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Adaptation of the rate is done in the end systems (gateways and end
users), and not in the routers.

If the problem you are concerned about is different MIPS at the end
systems (and thus varying abilities to support different codecs), this
is best handled during call setup through the various negotiation
protocols (SIP/SDP and H.323). 

There is the much harder problem, however, of adapting the rate at end
systems based on network QoS. There is no RFC which mandates any kind of
rate adaptation. There was a BoF called ADAPTS sometime back which was
looking at doing perhaps an informational RFC on the subject, but I
don't think anything came of it. I believe this to be an area where
there can be significant product/vendor differentiation.

The basic problem can be broken into a few parts:

1. the gateway must get information on what the end to end QoS is like
(in terms of delay, loss, jitter)

2. the gateway must decide what rate to send media at based on this
feedback

3. the gateway must change the rate of the media to meet the target in
(2).

Step (1) above is done using RTCP (the Real Time Control Protocol), part
of the RTP (rfc1889) specification. RTCP provides feedback from
receivers on delay, loss, and jitter. This information can be used by
applications to do step (2).

There are existing papers which cover (2), see in particular:

Busse, Deffner, Schulzrinne, "Dynamic QoS Control of Multimedia
Applications based on RTP", in the 1st International Workshop on High
Speed networks and Open Distributed Plaitforms, 1995

Bolot, Garcia, "Control Mechanisms for Packet Audio in the Internet",
Infocom 96

There is other work on emulating TCP behavior, but I don't have a
specific reference (anyone?). 

Step (3) comes into the realm of codecs and source coding.

 
-Jonathan R.


Wilson C. Chung wrote:
> 
> Hi Guys:
> 
> I am wondering is there any RFC that specifies "Traffic Shaping" in
> gateway(s) particularly for voice and video traffic.  I am thinking
> more or less an "intelligent network" that change the rate(s) and
> transcode various protocols in audio and video in either gateway(s)
> or router(s), with some regards to the End-to-End QoS client/server
> negotiation.  There is a big problem in a very heterogenous network
> where all sort of clients (in terms of MIPS) co-exist.
> 
> regards, Wilson
> 
> Wilson C. Chung, VM Labs
> wchung@vmlabs.com

-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX:   (732) 834-5379                       Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Thu Jan 28 07:18:08 1999 
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From: "Gio tesista" <valchesi70@hotmail.com>
To: rem-conf@es.net
Subject: multicast
Date: Thu, 28 Jan 1999 07:13:55 PST
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I 'm italian student and I 'm making  some experiments on multicast.
I have Linux configured 3 PC with multicast support and one of these  
is a multicast router(mrouter).If we create a work session on our same  
network,when we run "SD" or "SDR" I see the session on other computers 
and it's O.K..The problem is ONLY mrouter see external session (by 
tunneling) out my network,  but the other 2 computers no.What can I do 
to resolve this problem?
Thanks  Gio'

______________________________________________________
Get Your Private, Free Email at http://www.hotmail.com



From rem-conf Thu Jan 28 07:45:08 1999 
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Hi all,

Another tool called mIR (multicast Interactive Radio) has been around for a
while now. It is currently being managed and developed by Roland Parviainen. It
is mostly written in Java (except the actual decoder) and supports mpg123 and
WinAmp (depending on the platform you are on). It also supports DJ-like
functions such as voting for the next song that should be played.

Read more about mIR at: http://www.cdt.luth.se/~rolle/mIR/

/Serge Lachapelle
Marratech AB

Orion Hodson wrote:

> <3.0.5.16.19990127212022.2f6736fc@shell7.ba.best.com>Ross Finlayson writes:
> > A PLEA: Has anyone added MP3 decoding to "rat"?  If not, this would make a
> > great 'class project'.  (FYI, there is 'open source' MP3 decoding software
> > available at <http://dorifer.heim3.tu-clausthal.de/~olli/mpg123/>; it may
> > bepossible to use this in "rat".)
>
> Mark Handley suggested a while back that we add the mpg123 code (or
> something similar) and it is just one of those things I have never got
> around to doing.  If someone is interested it should be reasonably
> straightforward to incorporate mpg123 decoder.  The sources for the
> new codec interfaces in RAT are not yet available publicly, but if you
> are interested send a note to rat-trap@cs.ucl.ac.uk and we'll forward
> the codec components and some sample code.
>
> If someone takes this on I'll add the hooks for RAT to read coded
> files for transmission so it can send and receive layer-III files.
>
> There also are hooks in the experimental version of RAT to the Windows
> ACM codecs specifically for exploiting the Fraunhoffer Layer-3 codec
> (and G.723.1 but there's a key needed for this).  The code is
> currently suffering from bit-rot, but should get fixed for the next
> release.
>
> cheers
> - Orion
>
> =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
> Networked Multimedia Research Group      Tel: ++(0)171 419 3704
> Department of Computer Science           Fax: ++(0)171 419 1397
> University College London, Gower Street
> London, WC1E 6BT, UK.
> -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-




From rem-conf Thu Jan 28 07:48:13 1999 
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>
> There is other work on emulating TCP behavior, but I don't have a
> specific reference (anyone?).
>

http://www.psc.edu/networking/tcp_friendly.html

Fran=E7ois




From rem-conf Thu Jan 28 08:04:51 1999 
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From: "Angel L. Mateo" <angel.mateo@rediris.es>
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El d=EDa Thu, 28 Jan 1999 07:13:55 PST  "Gio tesista" escribi=F3:
> I 'm italian student and I 'm making  some experiments on multicast.
> I have Linux configured 3 PC with multicast support and one of these  =

> is a multicast router(mrouter).If we create a work session on our same =
 =

> network,when we run "SD" or "SDR" I see the session on other computers =

> and it's O.K..The problem is ONLY mrouter see external session (by =

> tunneling) out my network,  but the other 2 computers no.What can I do =

> to resolve this problem?

	I think that you have to a route to the mrouter in the other computers. =
For example, add:

	route add -net 224.0.0.0 netmask 240.0.0.0 dev eth0

to the /etc/rc.d/rc.local file


Salu2

Angel





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>> I 'm italian student and I 'm making  some experiments on multicast. 
I have Linux configured 3 PC with multicast support and one of these   
is a multicast router(mrouter).If we create a work session on our same 
 network,when we run "SD" or "SDR" I see the session on other computers  
and it's O.K..The problem is ONLY mrouter see external session (by  
tunneling) out my network,  but the other 2 computers no.What can I do  
to resolve this problem?
>
>	I think that you have to a route to the mrouter in the other 
computers. =
>For example, add:
>
>	route add -net 224.0.0.0 netmask 240.0.0.0 dev eth0
>
>to the /etc/rc.d/rc.local file
>
>
>Salu2
>
>Angel
>
>Thanks,but there is alredy this line .Infact if I run "route",I see 
also
>

Destination     Gatewey      Genmask    Flag  metric  Ref   Use  Iface
 
..........       .......              ......       .........   .
..........    ...........     ............
224.0.0.0         *        240.0.0.0      U     0      0     17   etho

Is it a problem di mrouter?I think no.What do I do?
Thanks.
          GIO,


______________________________________________________
Get Your Private, Free Email at http://www.hotmail.com



From rem-conf Thu Jan 28 16:18:24 1999 
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Reply-To: "Wilson C. Chung" <wchung@vmlabs.com>
From: "Wilson C. Chung" <wchung@vmlabs.com>
To: "Jonathan Rosenberg" <jdrosen@dnrc.bell-labs.com>
Cc: <rem-conf@es.net>
Subject: Re: Traffic Shaping
Date: Thu, 28 Jan 1999 16:05:41 -0800
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On part Three of the problem, I assumed you have to go
into the application layer and actually transcode the A/V
bitstreams.   Assumption number One is that bitstreams
are not scalable in rate nor resolution, therefore we have
to decode and encode again at gateway(s).    Most of
the broadcast and multimedia materials exist in the info
structure aren't ready scalable in layers either.  This will
push the MIPS requirement on a full "intelligent gateway"
concept infeasible for a "true" multicast scenerio.

Any thoughts?

Wilson Chung, VM Labs

-----Original Message-----
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
To: Wilson C. Chung <wchung@vmlabs.com>
Cc: rem-conf@es.net <rem-conf@es.net>
Date: Thursday, January 28, 1999 7:06 AM
Subject: Re: Traffic Shaping


>Adaptation of the rate is done in the end systems (gateways and end
>users), and not in the routers.
>
>If the problem you are concerned about is different MIPS at the end
>systems (and thus varying abilities to support different codecs), this
>is best handled during call setup through the various negotiation
>protocols (SIP/SDP and H.323).
>
>There is the much harder problem, however, of adapting the rate at end
>systems based on network QoS. There is no RFC which mandates any kind of
>rate adaptation. There was a BoF called ADAPTS sometime back which was
>looking at doing perhaps an informational RFC on the subject, but I
>don't think anything came of it. I believe this to be an area where
>there can be significant product/vendor differentiation.
>
>The basic problem can be broken into a few parts:
>
>1. the gateway must get information on what the end to end QoS is like
>(in terms of delay, loss, jitter)
>
>2. the gateway must decide what rate to send media at based on this
>feedback
>
>3. the gateway must change the rate of the media to meet the target in
>(2).
>
>Step (1) above is done using RTCP (the Real Time Control Protocol), part
>of the RTP (rfc1889) specification. RTCP provides feedback from
>receivers on delay, loss, and jitter. This information can be used by
>applications to do step (2).
>
>There are existing papers which cover (2), see in particular:
>
>Busse, Deffner, Schulzrinne, "Dynamic QoS Control of Multimedia
>Applications based on RTP", in the 1st International Workshop on High
>Speed networks and Open Distributed Plaitforms, 1995
>
>Bolot, Garcia, "Control Mechanisms for Packet Audio in the Internet",
>Infocom 96
>
>There is other work on emulating TCP behavior, but I don't have a
>specific reference (anyone?).
>
>Step (3) comes into the realm of codecs and source coding.
>
>
>-Jonathan R.
>
>
>Wilson C. Chung wrote:
>>
>> Hi Guys:
>>
>> I am wondering is there any RFC that specifies "Traffic Shaping" in
>> gateway(s) particularly for voice and video traffic.  I am thinking
>> more or less an "intelligent network" that change the rate(s) and
>> transcode various protocols in audio and video in either gateway(s)
>> or router(s), with some regards to the End-to-End QoS client/server
>> negotiation.  There is a big problem in a very heterogenous network
>> where all sort of clients (in terms of MIPS) co-exist.
>>
>> regards, Wilson
>>
>> Wilson C. Chung, VM Labs
>> wchung@vmlabs.com
>
>--
>Jonathan D. Rosenberg                       Lucent Technologies
>Member of Technical Staff                   101 Crawfords Corner Rd.
>High Speed Networks Research                Holmdel, NJ 07733
>FAX:   (732) 834-5379                       Rm. 4C-526
>EMAIL: jdrosen@bell-labs.com
>URL: http://www.cs.columbia.edu/~jdrosen





From rem-conf Thu Jan 28 17:24:27 1999 
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To: Ross Finlayson <finlayson@live.com>
cc: mbone@isi.edu, rem-conf@es.net
Subject: Re: A new tool for multicast MP3 audio streaming 
In-reply-to: Your message of "Wed, 27 Jan 99 13:20:22 PST."
             <3.0.5.16.19990127212022.2f6736fc@shell7.ba.best.com> 
Date: Thu, 28 Jan 1999 17:17:34 PST
Sender: Bill Fenner <fenner@parc.xerox.com>
From: Bill Fenner <fenner@parc.xerox.com>
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In message <3.0.5.16.19990127212022.2f6736fc@shell7.ba.best.com> Ross Finlayson wrote:
>- I doesn't automatically create SDP announcements
>    (instead, you'll need to announce your sessions 'by hand')

In order to facilitate creating these announcements, sdr users can
install this plugin as ~/.sdr/plugins/sdr2.plugin.mpegaudio .  It
won't allow launching (it will launch /bin/true) but it will allow
creation of the session with the MPEG Audio media type.

  Bill

#
# sdr2.plugin.mpegaudio
# Allows creation of sessions with the MPEG Audio media type.
media:		audio
proto:		RTP/AVP
tool:		true
protoname:	RTP

fmt: 14
{
	fmtname: MPEG Audio
}



From rem-conf Thu Jan 28 19:33:45 1999 
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From: "Neal Schneider" <nschneid@hotmail.com>
To: jdrosen@dnrc.bell-labs.com, wchung@vmlabs.com
Cc: rem-conf@es.net
Subject: Re: Traffic Shaping
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Do you have a source for this presentation?

>Bolot, Garcia, "Control Mechanisms for Packet Audio in the Internet",
>Infocom 96
>

>There is other work on emulating TCP behavior, but I don't have a
>specific reference (anyone?). 

Are you talking about providing some sort of acknowledgement-based 
negotiation protocol over UDP ... Does MGCP provide a mechanism for 
modifying connections in progress?

Regards, 
Neal A. Schneider 

>> Hi Guys:
>> 
>> I am wondering is there any RFC that specifies "Traffic Shaping" in
>> gateway(s) particularly for voice and video traffic.  I am thinking
>> more or less an "intelligent network" that change the rate(s) and
>> transcode various protocols in audio and video in either gateway(s)
>> or router(s), with some regards to the End-to-End QoS client/server
>> negotiation.  There is a big problem in a very heterogenous network
>> where all sort of clients (in terms of MIPS) co-exist.
>> 
>> regards, Wilson
>> 
>> Wilson C. Chung, VM Labs
>> wchung@vmlabs.com
>
>-- 
>Jonathan D. Rosenberg                       Lucent Technologies
>Member of Technical Staff                   101 Crawfords Corner Rd.
>High Speed Networks Research                Holmdel, NJ 07733
>FAX:   (732) 834-5379                       Rm. 4C-526
>EMAIL: jdrosen@bell-labs.com
>URL: http://www.cs.columbia.edu/~jdrosen
>


______________________________________________________
Get Your Private, Free Email at http://www.hotmail.com



From rem-conf Thu Jan 28 20:05:56 1999 
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Neal Schneider wrote:
> 
> Do you have a source for this presentation?
> 
> >Bolot, Garcia, "Control Mechanisms for Packet Audio in the Internet",
> >Infocom 96
> >

ftp://ftp-sop.inria.fr/rodeo/bolot/96.Audio_ctl.ps.gz

> 
> >There is other work on emulating TCP behavior, but I don't have a
> >specific reference (anyone?).
> 
> Are you talking about providing some sort of acknowledgement-based
> negotiation protocol over UDP

No. I am talking about rate shaping in such a way that the rate used by
the media stream is the same rate a TCP stream would use under the same
loss conditions. 

 ... Does MGCP provide a mechanism for
> modifying connections in progress?

MGCP can modify its current connections. However, I don't know if rate
adaptation is something the protocol is aiming to control. I'm also not
sure rate adaptation is best done in the MGC, since its a pure media
function it could be done entirely in the MG.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX: (732) 834-5379                         Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Thu Jan 28 20:27:23 1999 
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Wilson C. Chung wrote:
> 
> On part Three of the problem, I assumed you have to go
> into the application layer and actually transcode the A/V
> bitstreams. Assumption number One is that bitstreams
> are not scalable in rate nor resolution, therefore we have
> to decode and encode again at gateway(s). 

No, the media arrives from the circuit uncompressed, so you need only
compress once at the ingress gateway. The adaptation is done (for voice,
at least), by codec selection since most codecs don't have tunable
parameters to control the rate. G.723.1 is a minor exception - you can
switch between 5.3 and 6.3 kbps. There is a lot more control for video
(adjusting things like quantization parameters), and perhaps for high
quality audio.



   Most of
> the broadcast and multimedia materials exist in the info
> structure aren't ready scalable in layers either.  This will
> push the MIPS requirement on a full "intelligent gateway"
> concept infeasible for a "true" multicast scenerio.

True, speech is not really amenable to layered coding since its so low
bitrate to begin with. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX: (732) 834-5379                         Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen



From rem-conf Thu Jan 28 20:39:11 1999 
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From: Reza Rejaie <reza@catarina.usc.edu>
To: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
cc: Neal Schneider <nschneid@hotmail.com>, wchung@vmlabs.com, rem-conf@es.net
Subject: Re: Traffic Shaping
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On Thu, 28 Jan 1999, Jonathan Rosenberg wrote:

> Neal Schneider wrote:
> > 
> > Do you have a source for this presentation?
> > 
> > >Bolot, Garcia, "Control Mechanisms for Packet Audio in the Internet",
> > >Infocom 96
> > >
> 
> ftp://ftp-sop.inria.fr/rodeo/bolot/96.Audio_ctl.ps.gz
> 
> > 
> > >There is other work on emulating TCP behavior, but I don't have a
> > >specific reference (anyone?).
> > 
> > Are you talking about providing some sort of acknowledgement-based
> > negotiation protocol over UDP
> 
> No. I am talking about rate shaping in such a way that the rate used by
> the media stream is the same rate a TCP stream would use under the same
> loss conditions. 

We have designed a TCP-friendly Rate Adaptation Protocol(RAP) for
streaming applications that achieves this goal.
For more details refer to :

"RAP: An End-to-end Rate-based Congestion Control Mechanism for Realtime 
Streams in the Internet" 
Reza Rejaie, Mark Handely and Deborah Estrin, 
To appear in IEEE Infocom'99 
http://netweb.usc.edu/reza/papers/infocom99.ps

ReZa



From rem-conf Fri Jan 29 03:43:35 1999 
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Date: Fri, 29 Jan 1999 03:34:51
To: mbone@isi.edu, rem-conf@es.net
From: Ross Finlayson <finlayson@live.com>
Subject: A MPEG/RTP audio tool for Windows
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Following up on my earlier message about liveCaster, I've hacked together a
Windows application that receives RTP-encapsulated multicast MPEG audio
packets, and plays the MPEG data using Nullsoft's "Winamp" audio player tool.

This application - "playRTPMPEG" - is available from
	<http://www.live.com/multikit/playRTPMPEG.html>

If you're using the "multikit" session browser, this application (once
installed) will be run automatically if you launch a MPEG audio (i.e.,
payload type 14) session.  Doing the same for other SDP browsers (e.g.,
"sdr") should be straightforward.

How it works:
It's basically a hack.  It works by receiving RTP-encapsulated MPEG data,
and streaming the decapsulated MPEG frames to Winamp using HTTP, i.e., by
acting as a special-purpose HTTP server.  (Thanks to Roland Parviainen for
this suggestion.)

It would have been straightforward to build Unix versions of this tool that
work the same way - e.g., by streaming to "mpg123".  I haven't done this,
though, because (i) I don't have sound on any of my Unix development
machines, and (ii) it's the wrong thing to do anyway.  Instead the right
thing to do would be to either (a) integrate a MPEG decoder into an
existing open RTP/RTCP-based audio player (such as "rat"), or (b) integrate
RTP/RTCP handling into an existing open MPEG player (such as "mpg123").

	Ross.

ps. BTW, I'm still not 100% sure whether or not liveCaster's implementation
of MPEG-audio-in-RTP is correct, because I have yet to test it with any
other receiving implementation.  If anyone out there has their own
RTP-capable MP3 player, please let me know whether liveCaster works (or
doesn't work) with it.




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From: "Gio tesista" <valchesi70@hotmail.com>
To: angel.mateo@rediris.es
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Subject: Re: multicast
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>>	I think that you have to a route to the mrouter in the other 
>computers. =
>>For example, add:
>>
>>	route add -net 224.0.0.0 netmask 240.0.0.0 dev eth0
>>
>>to the /etc/rc.d/rc.local file
>>
>>
>>Salu2
>>
>>Angel
>>

Thanks,but there is alredy this line .Infact if I run "route",I see 
also


>Destination     Gatewey      Genmask    Flag  metric  Ref   Use  Iface
> 
>..........       .......              ......       .........   .
>..........    ...........     ............
>224.0.0.0         *        240.0.0.0      U     0      0     17   etho
>
>Is it a problem di mrouter?or is it a problem of setting TTL in 
mrouter? What can I do?
>Thanks.
>          GIO,
>
>

>>> I 'm italian student and I 'm making  some experiments on multicast. 
>I have Linux configured 3 PC with multicast support and one of these   
>is a multicast router(mrouter).If we create a work session on our same 
> network,when we run "SD" or "SDR" I see the session on other computers  
>and it's O.K..The problem is ONLY mrouter see external session (by  
>tunneling) out my network,  but the other 2 computers no.What can I do  
>to resolve this problem?
>>

>______________________________________________________
>Get Your Private, Free Email at http://www.hotmail.com
>
>


______________________________________________________
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From rem-conf Sat Jan 30 10:40:28 1999 
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	id 106fA9-0001oy-00; Sat, 30 Jan 1999 10:30:36 -0800
From:     BMC  (USA) <bmc_mailing@bigfoot.com>
To:        <rem-conf@es.net>
Message-Id: <419.436190.56198310bmc_mailing@bigfoot.com>
Subject: ADV: Mail Forwarding Service 
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Date: Sat, 30 Jan 1999 10:30:36 -0800
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Dear Reader,

This is a commercial message offering a mail forwarding service. If you wish to be 
removed from 
this mailing list, reply with "Remove" in the subject line, and you will be deleted from 
our list.  We 
do not sell or share our mailing lists.

MAIL FORWARDING SERVICE (ESTABLISHED 1995)

Enjoy the benefits of having a real street address for the reception of your business or 
personal 
mail.  We guarantee total confidentiality and security. Mail forwarded to you according 
to your 
instructions.

Ideal for: 
- traveling businessmen,
- students,
- direct marketers,
- non USA residents,
- importers/exporters,
- and others.
 
Benefits:
 - confidential
 - cost effective

MAIL RECEIVING 
We can receive letters and parcels from the US Postal Service and, unlike a P.O. Box, 
we can 
also recieve shipments from U.P.S., FedEx, DHL, and others. You can also receive 
registered 
and certified mail, should you authorize us to do so.

Our mail receiving service provides you with a legitimate New Jersey street address, 
not a P.O. 
Box. This provides many benefits for individuals and businesses alike. Addresses in 
other areas 
are not available at this time.

SECURITY 
We are specialists in the rental of private addresses, packaging and shipping 
services. We also 
adhere to the highest standards of confidentiality and security. Your mail is handled in 
a private, 
secure fashion by experienced professionals. The confidentiality of mail processing 
is our number 
one concern. 

MAIL FORWARDING SERVICE FEES
We can redirect your mail anywhere in the world, according to the your instructions.

The handling charges to redirect mail are:
$0.50 per letter (business size envelope) plus postage/shipping costs
$5.00 for boxed parcels plus postage/shipping costs

The mail is forwarded to you according to your instructions, either daily, weekly or 
monthly.

The address would look like this (example):

(Your Name) or (Care of BM&Co.)
712 Wyoming Ave. 
Elizabeth,  NJ  07208
USA

RATES
$10.00 USD per month (minimum rental is 6 months)

Example:  Rental for 6 months would cost:
                $  60.00 USD (Rental) 
                $  50.00 USD (Deposit for Postage/Handling to redirect your mail)
  Total       $110.00 USD

Our rates are among the lowest in the industry and our service is among the best, and 
unlike 
others, we have absolutely no setup fees.

To sign up for our mail forwarding service, please mail us:

- Your name;
- Address where you want your mail forwarded to;
- Copy of photo ID;
- Your e mail address;
- Any special instructions;
- Phone or fax number (optional).

We accept payment by check or money order, in US dollars. We do not accept credit 
cards. Also 
inform us of any special arrangements for the handling of your mail.

Upon receipt of payment, you will be sent your address via e mail.


I look forward to hearing from you. If you have any questions, please call between 9 
am to 6pm 
USA eastern time. At other hours, please leave a message and we will return your 
call.

Sincerely,

Bruno Teixeira
Accounts Manager

BMC  (USA)
bmc_mailing@bigfoot.com
Tel: 00 1 908 820 0720
Fax: 00 1 908 820 4972





From rem-conf Sun Jan 31 00:50:21 1999 
From rem-conf-request@es.net Sun Jan 31 00:50:20 1999
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	id 106sU1-000687-00; Sun, 31 Jan 1999 00:43:57 -0800
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	id 106sTy-00067x-00; Sun, 31 Jan 1999 00:43:54 -0800
From:     Telco Independent Agent <TimBrooks4@aol.com>
To:        <rem-conf@es.net>
Message-Id: <419.436191.36541655TimBrooks4@aol.com>
Subject: Save Money on UK Telephone Calls
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Date: Sun, 31 Jan 1999 00:43:54 -0800
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Please take a second to learn how to SAVE 66% on BT residential bills, 80% on 
Orange or 1-2-1 
call charges using the Telco freephone access service.... .

- After free registration, simply dial a freephone access number then the destination 
number to 
save money on call charges
- Free Monthly Itemised Billing
- NO deposits, NO line rental, NO operators, NO new equipment

For further informaiton please visit:-

www.angelfire.com/co/telcoinfo/index.html

Thank You




From rem-conf Sun Jan 31 03:01:15 1999 
From rem-conf-request@es.net Sun Jan 31 03:01:14 1999
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	id 106uX3-0007QZ-00; Sun, 31 Jan 1999 02:55:13 -0800
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	id 106uX1-0007QP-00; Sun, 31 Jan 1999 02:55:11 -0800
To: <rem-conf@es.net>
From: <bunniesam@eastmail.com>
Subject: PUT MONEY INTO YOUR POCKET!  BUSINESS OPPORTUNITY AD 
Date: Wed, 3 Feb 1999 09:17:17
Message-Id: <E106uX1-0007QP-00@mail1.es.net>
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                 VISA/MASTERCARD/AMEX/DISCOVER

                     STOP LOSING MONEY

                  ACCEPT CREDIT CARDS TODAY

67% OF YOUR COMPETITION ACCEPTS CREDIT CARDS.  DON'T LET YOUR 
COMPETITION STEAL YOUR CUSTOMERS.

                       RISK FREE OFFER

NO EQUIPMENT                               NO LEASE

                         NO EXCUSES

E COMMERCE AND TELEPHONE PROCESSING

CALL STEVE MILLER 1888-290-0617


THIS MESSAGE SENT BY EMAIL  KING AND ASSOCIATES
1790 BONHILL RD
MISSISSAUGA ON
905 564 5091

TO BE REMOVED DIAL 800 636-6773 X 4692 OR EMAIL 
bunniesam@EASTMAIL.COM


 
 
 
 
 
 
 
 
 
 
 
 
 



From rem-conf Sun Jan 31 19:58:53 1999 
From rem-conf-request@es.net Sun Jan 31 19:58:53 1999
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	id 107ANS-0006Ey-00; Sun, 31 Jan 1999 19:50:22 -0800
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	id 107ANP-0006Eo-00; Sun, 31 Jan 1999 19:50:20 -0800
From: <acctsvs@bellsouth.net>
Subject: ADV:CREDIT CARD PROCESSING
Date: Sun, 31 Jan 1999 16:12:17
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*********************************************************
This message is being brought to you by World Tek Inc.
To be removed from from further mailings respond to this
message with "remove" in the subject line.
*********************************************************

Dear Friend,

Discover how you can accept credit cards directly
>from your website, telephone or fax for your products 
and services and never need to purchase or lease
expensive credit card equipment or pay a large monthly 
fee for online ordering capabilities or real time processing
transactions.

**Brand New**  Inteli-Charge Credit Card acceptance program 
allows you to accept Visa, MastercardTM, Amex and Discover 
any TIME,any WHERE through phone, fax or internet without 
the need to purchase or lease expensive credit card equipment. 
This brand new program will allow you to accept credit cards 
in 24-48 hours after submitting your application.

You simply pick up your telephone, dial a special toll free
800# 24 hrs a day 7 days a week, input a passcode and the
credit card # and receive an immediate authorization over the 
phone.
Within 2 days the money is deposited in your bank account. This 
is an exciting program for all businesses. Before you spend any 
money on a credit card merchant program LOOK at this new program! 
We have virtually a 100% approval for most business types 
regardless of past credit history!


If you have an interest in learning more about a Merchant Account 
for yourself or your business please email your Name, PHONE 
NUMBER (Don't forget your area code) and best time to call to:

mailto:acctsvs@bellsouth.net
 
A representative will return your call within 24hrs.

Or feel free to call us on our 24 hour voicemail at:

1-800-600-0343 Ext:2104

P.S. ALSO AS PART OF A SPECIAL NATIONAL PROMOTION, IF
YOU ACT NOW YOU WILL RECEIVE A FREE ONLINE 
STORE WITH SECURE ORDERING CAPABILITIES
TO SELL YOUR PRODUCTS ONLINE! 

This offer only applies to U.S. Residents only and some Canadians 
with valid U.S. Social Security #'s

World Tek Inc.
7210 Jordan Ave.
Canoga Park Ca. 91303

 
 
 
 
 
 
 
 
 



