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As per the WG chairs' requests, I made a revision on RFC 2250. However, I =
don't have the current coordinates of my co-authors. So, Don Hoffman and =
Gerard Fernando, if you are still observing this list please let me know =
your coordinates. Rest of the list, please accept my apologies for this.




------------------------------------------------------
M. Reha Civanlar
Computer Engineering Department
Ko=E7 University
Istanbul, TURKEY
+90 212 338 1719
rcivanlar@ku.edu.tr


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3550

        Title:      RTP: A Transport Protocol for Real-Time
                    Applications
        Author(s):  H. Schulzrinne, S. Casner,
                    R. Frederick, V. Jacobson
        Status:     Standards Track
        Date:       July 2003
        Mailbox:    schulzrinne@cs.columbia.edu, casner@acm.org,
                    ronf@bluecoat.com, van@packetdesign.com
        Pages:      89
        Characters: 630740
        Obsoletes:  1889

        I-D Tag:    draft-ietf-avt-rtp-new-12.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3550.ps


This memorandum describes RTP, the real-time transport protocol.  RTP
provides end-to-end network transport functions suitable for
applications transmitting real-time data, such as audio, video or
simulation data, over multicast or unicast network services.  RTP
does not address resource reservation and does not guarantee
quality-of-service for real-time services.  The data transport is
augmented by a control protocol (RTCP) to allow monitoring of the
data delivery in a manner scalable to large multicast networks, and
to provide minimal control and identification functionality.  RTP and
RTCP are designed to be independent of the underlying transport and
network layers.  The protocol supports the use of RTP-level
translators and mixers.

Most of the text in this memorandum is identical to RFC 1889 which it
obsoletes.  There are no changes in the packet formats on the wire,
only changes to the rules and algorithms governing how the protocol
is used.  The biggest change is an enhancement to the scalable timer
algorithm for calculating when to send RTCP packets in order to
minimize transmission in excess of the intended rate when many
participants join a session simultaneously.

This document is a product of the Audio/Video Transport Working Group
of the IETF.

This is now a Draft Standard Protocol.

This document specifies an Internet standards track protocol for the
Internet community, and requests discussion and suggestions for
improvements.  Please refer to the current edition of the "Internet
Official Protocol Standards" (STD 1) for the standardization state
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Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

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_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3551

        Title:      RTP Profile for Audio and Video Conferences
                    with Minimal Control
        Author(s):  H. Schulzrinne, S. Casner
        Status:     Standards Track
        Date:       July 2003
        Mailbox:    schulzrinne@cs.columbia.edu, casner@acm.org
        Pages:      38
        Characters: 317286
        Obsoletes:  1890

        I-D Tag:    draft-ietf-avt-profile-new-13.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3551.ps


This document describes a profile called "RTP/AVP" for the use of the
real-time transport protocol (RTP), version 2, and the associated
control protocol, RTCP, within audio and video multiparticipant
conferences with minimal control.  It provides interpretations of
generic fields within the RTP specification suitable for audio and
video conferences.  In particular, this document defines a set of
default mappings from payload type numbers to encodings.

This document also describes how audio and video data may be carried
within RTP.  It defines a set of standard encodings and their names
when used within RTP.  The descriptions provide pointers to reference
implementations and the detailed standards.  This document is meant
as an aid for implementors of audio, video and other real-time
multimedia applications.

This memorandum obsoletes RFC 1890.  It is mostly backwards-compatible
except for functions removed because two interoperable implementations
were not found.  The additions to RFC 1890 codify existing practice in
the use of payload formats under this profile and include new payload
formats defined since RFC 1890 was published.

This document is a product of the Audio/Video Transport Working Group
of the IETF.

This is now a Draft Standard Protocol.

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Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

...

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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Retransmission Payload Format
	Author(s)	: J. Rey et al.
	Filename	: draft-ietf-avt-rtp-retransmission-09.txt
	Pages		: 27
	Date		: 2003-8-5
	
RTP retransmission is an effective packet loss recovery technique 
for real-time applications with relaxed delay bounds.  This 
document describes an RTP payload format for performing 
retransmissions.  Retransmitted RTP packets are sent in a separate 
stream from the original RTP stream.  It is assumed that feedback 
from receivers to senders is available.  In particular, it is 
assumed that RTCP feedback as defined in the extended RTP profile 
for RTCP-based feedback (denoted RTP/AVPF), is available in this 
memo.

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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] NOMS 2004 - Nine days to the Deadline - 15th August 2003
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***** Paper submission deadline extended to Friday 15th August 2003 *****

        Network Operation & Management Symposium 2004 

"Managing Next Generation Convergence Networks and Services"

                Seoul Korea, 19-23, April 2004

See < http://www.noms2004.org/> for latest information

The 9th IEEE/IFIP Network Operations and Management Symposium (NOMS 2004)
will be held 19-23 April, 2004 at Seoul, Korea. Held in the even-numbered
years since 1988, NOMS 2004 will continue the established tradition of
NOMS and IM as the primary forum for technical exchange of research,
standards, development, systems integrator, service providers, and user
communities. NOMS 2004 will present the latest approaches and technical
solutions in the area of network operations and management. An exciting,
peer-reviewed program of technical sessions, tutorials, posters, panels
and vendor exhibits will address the ever-increasing interest in overall
management solutions for all types of communications and computing
networks, systems, services and enterprise applications.

The concept of network convergence has recently emerged as a new attempt
for the merging of telephony and data networks into a single,
multi-service network exploiting the ubiquity of the Internet Protocol.
For this increasingly attractive business model, in both wired and
wireless domains, strategic research is required to devise the best
integration architectures, operations and management solutions. This
creates a unique opportunity for the network operation and management
community to respond to the ever-increasing demand for network resilience,
security, quality-of-service and mobility management at unprecedented
scales. The NOMS 2004 provides the forum for discussing these research
challenges and many others inherent to the integrated management of next
generation converged networks and services.

This year NOMS 2004 will broaden the scope of previous IM and NOMS by
expanding its program to include a broader set of topics ranging from
network operation and management to network planning, service engineering
and business processes for network and service management. Special
attention will be given to experiences that emphasize lessons learned and
reports on practice from industry. Authors are invited to submit complete
unpublished papers, which are not under review in any other conference or
journal. Authors are also invited to submit proposals for tutorials, panel
discussions, poster demonstrations, or birds-of-a-feather sessions.

Important Dates:
Deadline  for  Submitting  Papers: 15  August   2003
Deadline  for  Tutorials, Panels and  Posters:  1 October  2003
Notification of Acceptance:15 November 2003
Deadline for Submitting Revised Papers:15 December 2003






_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Sun Aug 10 22:57:50 2003
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Hello;

   Can anyone tell me what has happened to 
draft-ietf-avt-rtcp-report-extns ,
RTCP reporting extensions ?

   Did this make RFC ? What is the current version ?


                                 Regards
                                 Marshall Eubanks

T.M. Eubanks
e-mail : tme@multicasttech.com
http://www.multicasttech.com

Test your network for multicast :
http://www.multicasttech.com/mt/

Our New Video Service is in Beta testing
http://www.americafree.tv

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Sun Aug 10 23:13:29 2003
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From: "Alan Clark" <alan@telchemy.com>
To: "Marshall Eubanks" <tme@multicasttech.com>, <avt@ietf.org>
Subject: RE: [AVT] RR Reporting Extensions ?
Date: Sun, 10 Aug 2003 23:16:53 -0400
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Marshall

Draft -06 was approved by the IESG and is now in the RFC Editor queue.  You
can track it's progress through the IESG section of the IETF web site.

Regards

Alan Clark


-----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of
Marshall Eubanks
Sent: Sunday, August 10, 2003 10:56 PM
To: avt@ietf.org
Subject: [AVT] RR Reporting Extensions ?


Hello;

   Can anyone tell me what has happened to
draft-ietf-avt-rtcp-report-extns ,
RTCP reporting extensions ?

   Did this make RFC ? What is the current version ?


                                 Regards
                                 Marshall Eubanks

T.M. Eubanks
e-mail : tme@multicasttech.com
http://www.multicasttech.com

Test your network for multicast :
http://www.multicasttech.com/mt/

Our New Video Service is in Beta testing
http://www.americafree.tv

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Mon Aug 11 04:16:49 2003
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From: "Jose Rey" <rey@panasonic.de>
To: "Colin Perkins" <csp@csperkins.org>
Cc: <avt@ietf.org>
Subject: RE: [AVT] 3GPP timed text draft: more generic?
Date: Mon, 11 Aug 2003 10:15:44 +0200
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Hi all,

--snip
Colin Perkins wrote:
> A specific payload format for 3GPP timed text might be
> appropriate, but I
> would like to understand the options and alternatives before
> we make that
> choice. Is 3GPP the only body doing timed and/or decorated
> text? If not,
> what are the alternatives, and how do they relate?
>
> --

at the moment there are at least two more standardisation bodies
interested
in timed text: MPEG and W3C.


- MPEG is currently defining their own TT format based on 3GPP TT (see
Committee Draft of ISO/IEC 14496-17 from July'03 Meeting in Trondheim)
by defining the access unit structure, the stream type, the object type
indication and decoder specific info of the timed text samples. The
definition of these items enables the transport over RTP using the
mpeg4simple draft. The work in MPEG is assumed to be finalised around
July 2004.

- W3C is at a very early stage in defining a timed text authoring
format. It
has recently (May'03) published a draft for use cases and requirements
of a
timed text authoring format:
http://www.w3.org/TR/2003/WD-tt-af-1-0-req-20030515/. This draft has a
rather long list of requirements and lists many functionalities. In my
understanding W3C targets to a generic format with rich functionality. I
don't know how quick it may develop (anyone?).

None of these above is an option for us, because 1) the only timed text
format in
3GPP is 3GPP timed text 2) 3GPP has not adopted the mpeg4simple for
transport and 3) for this 3GPP timed text format we need an RTP payload
format within the current 3GPP Release 6 schedule (i.e. March'04).


cheers,

Jose




> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]
> Sent: Monday, July 28, 2003 3:18 PM
> To: Jose Rey
> Cc: avt@ietf.org
> Subject: Re: [AVT] 3GPP timed text draft: more generic?
>
>
> --> "Jose Rey" <rey@panasonic.de> writes:
> >
> > Hi all,
> >
> > during the presentation of the payload format for 3GPP
> timed text some
> > concerns arose on whether this work should be more generic.  On the
> > other hand I also emphasized the fact that there's a
> deadline to meet
> > here for 3GPP Release 6: March '04.
> >
> > It is our opinion that, in this case, a more specific
> payload format for
> > the 3GPP timed text format is the right way to go. Does
> anybody think
> > otherwise? We would like to hear comments/critic on this,
> in order to
> > progress to working group work item.  How could it be made
> more generic?
>

> Colin Perkins
> csp@csperkins.org


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] Call for Papers: WiOpt'04
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Apologies if you receive multiple copies of this call!

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

Note:  The submission deadline is 5th of November, 2003.


         ***************************************************
         ********          CALL FOR PAPERS          ********
         ***************************************************
         *                                                 *
         *                   WiOpt '04                     *
         *                                                 *
         *     2nd Workshop on Modeling and Optimization   *
         *      in Mobile, Ad Hoc and Wireless Networks    *
         *                                                 *
         *        March 24 - 26, 2004, Cambridge, UK       *
         *                                                 *
         *                                                 *
         ***************************************************


                          Workshop Website:
            http://www.cl.cam.ac.uk/Research/SRG/wiopt04/


This workshop is intended to bring together leading researchers from
both academia and industry to present both foundations and practical
methods and tools for modeling and optimisation of mobile and ad hoc
networks.  The workshop will bring together researchers studying
optimisation issues of mobile networks from both the network
performance and the mobile networking perspectives:

*  Mobile Network Optimisation: Applications and Algorithms
*  Performance characterisation and optimisation of
   mobile/wireless networks and systems
*  Protocol design (MAC, Routing) for optimal mobile network
   utilisation
*  Tools for network optimisation
*  Energy efficiency in mobile networks
*  Security and co-operation models for mobile networks
*  Pricing and incentives in mobile and ad hoc networks.

Papers to be presented at WiOpt'04 will be selected based on extended
abstracts of up to 8 pages, double-spaced, 12 point font size. The
submission should not exceed 3000 words including tables and figures,
each of which count as 250 words.  Authors of accepted presentations
will have the opportunity to publish either the extended abstract or
the full paper in a proceedings volume. Full papers will further be
considered for publication in a special issue of the MONET journal. 

Plenary Speakers:

Bruce Hajek (University of Illinois at Urbana-Champaign, USA) 
Ian Akyildiz (Georgia Institute of Technology, USA) 


IMPORTANT DATES:
Submission of extended abstract: Nov.  5, 2003.
Notification of acceptance:      Jan. 15, 2003.
Camera ready copy:               Feb. 15, 2003.

(Further details on the workshop website,
http://www.cl.cam.ac.uk/Research/SRG/wiopt04/)



General chairman: Imrich Chlamtac

Vice General Chair: Roger Whitaker 

Program Co-Chairs: Marco Conti and Tamer Basar

Program committee of the conference:

Eitan Altman, Venkat Anantharam, Konstantin Avrachenkov, Andrew
Campbell, A. Chockalingam, Jon Crowcroft, Silvia Giordano, Stephen
V. Hanly, Jean-Pierre Hubaux, Holger Karl, Peter Key, P. R. Kumar,
Peter Marbach, Philippe Nain, Sven Oestring, Yoshikuni Onozato, Ramesh
Rao, Ness Shroff, Violet Syrotiuk, Thierry Turletti, Adam Wolisz, Jeff
Wieselthier

Local Arrangements Co-Chair: Sven Oestring and Julian Chesterfield

Publicity Chair: Holger Karl


Sponsors:
The workshop is co-sponsored by
        * Univ. of Cambridge
        * IIT (Istituto di Informatica e Telematica)
        * Intel
        * Microsoft
        * IEEE IT Society (technical co-sponsorship)
(ACM sponsorship is pending.)



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Mon Aug 11 05:36:29 2003
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Date: Mon, 11 Aug 2003 10:34:33 +0100
Subject: Re: [AVT] Working group last call: draft-ietf-avt-rfc2833bis-03.txt
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On Monday, Jul 28, 2003, at 02:15 Europe/London, Colin Perkins wrote:
> This is to announce a working group last call on the RTP Payload for 
> DTMF
> Digits, Telephony Tones and Signals <draft-ietf-avt-rfc2833bis-03.txt>.
> Please send any final comments on this draft to the mailing list by 
> 11th
> August 2003. If no substantive technical issues are raised, we intend 
> to
> submit this draft for publication as a proposed standard RFC to 
> obsolete
> RFC 2833.
>
> Note that this draft will not be advanced until we receive at least one
> valid review from someone not directly involved with production of the
> draft (i.e., not an author or contributor).

This is to solicit reviews of the draft-ietf-avt-rfc2833bis-03.txt. As 
noted above, we will not advance the draft until at least one valid 
technical review, from someone not directly involved with the 
production of the draft, is received.

-- 
Colin Perkins
http://csperkins.org/


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Mon Aug 11 06:04:25 2003
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Subject: Re: [AVT] 3GPP timed text draft: more generic?
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On Monday, Aug 11, 2003, at 09:15 Europe/London, Jose Rey wrote:
> Colin Perkins wrote:
>> A specific payload format for 3GPP timed text might be
>> appropriate, but I
>> would like to understand the options and alternatives before
>> we make that
>> choice. Is 3GPP the only body doing timed and/or decorated
>> text? If not,
>> what are the alternatives, and how do they relate?
>>
>> --
>
> at the moment there are at least two more standardisation bodies
> interested in timed text: MPEG and W3C.
>
> - MPEG is currently defining their own TT format based on 3GPP TT (see
> Committee Draft of ISO/IEC 14496-17 from July'03 Meeting in Trondheim)
> by defining the access unit structure, the stream type, the object type
> indication and decoder specific info of the timed text samples. The
> definition of these items enables the transport over RTP using the
> mpeg4simple draft. The work in MPEG is assumed to be finalised around
> July 2004.
>
> - W3C is at a very early stage in defining a timed text authoring
> format. It has recently (May'03) published a draft for use cases and 
> requirements of a timed text authoring format:
> http://www.w3.org/TR/2003/WD-tt-af-1-0-req-20030515/. This draft has a
> rather long list of requirements and lists many functionalities. In my
> understanding W3C targets to a generic format with rich functionality. 
> I
> don't know how quick it may develop (anyone?).

I believe SMPTE may also have timed text work in their Digital Cinema 
group?

> None of these above is an option for us, because 1) the only timed text
> format in 3GPP is 3GPP timed text 2) 3GPP has not adopted the 
> mpeg4simple for
> transport and 3) for this 3GPP timed text format we need an RTP payload
> format within the current 3GPP Release 6 schedule (i.e. March'04).

I agree that using the MPEG4-simple format doesn't make sense, and that 
the W3C schedule looks to be too long for 3GPP requirements.

-- 
Colin Perkins
http://csperkins.org/


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Payload Format for MPEG1/MPEG2
	Author(s)	: D. Hoffman et al.
	Filename	: draft-ietf-avt-mpeg1and2-mod-00.txt
	Pages		: 15
	Date		: 2003-8-12
	
This memo describes a packetization scheme for MPEG video and audio 
streams.  The scheme proposed can be used to transport such a video 
or audio flow over the transport protocols supported by RTP.  Two 
approaches are described. The first is designed to support maximum 
interoperability with MPEG System environments.  The second is 
designed to provide maximum compatibility with other RTP-encapsulated 
media streams and future conference control work of the IETF. 
Most of this memo is identical to RFC 2250, an Internet standards 
track RTP payload format definition. No changes have been made in the 
packet formats on the wire. The main reason for this revision is to 
allow the use of this payload format with dynamic payload types  
that can specify the timestamp clock frequency by non-RTP means for 
improved jitter compensation. We used this opportunity to improve the 
description of the payload format specification by clarifying some 
wording that have been reported to be problematic.

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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] [Fwd: I-D ACTION:draft-jones-avt-audio-t38-00.txt]
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What's this all about?  First point is that image/t38 is already a registered MIME 
type.  Second point is that T.38 does NOT use RTP transport, in contradiction to the 
statement in the I-D that it does.

-------- Original Message --------
Subject: I-D ACTION:draft-jones-avt-audio-t38-00.txt
Date: Mon, 11 Aug 2003 15:46:16 -0400
From: Internet-Drafts@ietf.org
Reply-To: Internet-Drafts@ietf.org
To: IETF-Announce: ;

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


	Title		: Real-Time Facsimile (T.38) - audio/t38 MIME Sub-type
                           Registration
	Author(s)	: P. Jones
	Filename	: draft-jones-avt-audio-t38-00.txt
	Pages		: 4
	Date		: 2003-8-11
	
This document defines the MIME sub-type audio/t38.  The packetization
and usage of this MIME type, which is intended for use within SDP [2]
and H.248 [3] systems, is specified within ITU-T Recommendation T.38
[1].

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-jones-avt-audio-t38-00.txt

To remove yourself from the IETF Announcement list, send a message to
ietf-announce-request with the word unsubscribe in the body of the message.

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-jones-avt-audio-t38-00.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
...


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Tue Aug 12 22:22:37 2003
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From: "Paul E. Jones" <paulej@packetizer.com>
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Tom,

The purpose is to register the MIME type "audio/t38".  The reason is that
work is underway within ITU-T Q14/16 to support the carriage of T.38's IFP
packets through RTP.  This will allow us to then signal a single "m=" like
that looks like this:

  m=audio 5600 RTP/AVP 18 98
  a=rtpmap:98 t38/8000

When two media gateways have negotiated this (or similar) SDP and a fax tone
is detected on the line side, a mode shift may occur by simply transmitting
packets with PT=98, rather than PT=18.  This reduces the need for either
consuming additional UDP ports per call for image/t38 or going through a
lengthy exchange of messages to open a fax channel.

Paul

----- Original Message ----- 
From: "Tom Taylor" <taylor@nortelnetworks.com>
To: <avt@ietf.org>
Cc: <paulej@packetizer.com>
Sent: Tuesday, August 12, 2003 11:10 AM
Subject: [Fwd: I-D ACTION:draft-jones-avt-audio-t38-00.txt]


> What's this all about?  First point is that image/t38 is already a
registered MIME
> type.  Second point is that T.38 does NOT use RTP transport, in
contradiction to the
> statement in the I-D that it does.
>
> -------- Original Message --------
> Subject: I-D ACTION:draft-jones-avt-audio-t38-00.txt
> Date: Mon, 11 Aug 2003 15:46:16 -0400
> From: Internet-Drafts@ietf.org
> Reply-To: Internet-Drafts@ietf.org
> To: IETF-Announce: ;
>
> A New Internet-Draft is available from the on-line Internet-Drafts
directories.
>
>
> Title : Real-Time Facsimile (T.38) - audio/t38 MIME Sub-type
>                            Registration
> Author(s) : P. Jones
> Filename : draft-jones-avt-audio-t38-00.txt
> Pages : 4
> Date : 2003-8-11
>
> This document defines the MIME sub-type audio/t38.  The packetization
> and usage of this MIME type, which is intended for use within SDP [2]
> and H.248 [3] systems, is specified within ITU-T Recommendation T.38
> [1].
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-jones-avt-audio-t38-00.txt
>
> To remove yourself from the IETF Announcement list, send a message to
> ietf-announce-request with the word unsubscribe in the body of the
message.
>
> Internet-Drafts are also available by anonymous FTP. Login with the
username
> "anonymous" and a password of your e-mail address. After logging in,
> type "cd internet-drafts" and then
> "get draft-jones-avt-audio-t38-00.txt".
>
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> ...
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Aug 13 04:54:42 2003
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From: "Karl Norrman (KI/EAB)" <karl.norrman@ericsson.com>
To: "'avt@ietf.org'" <avt@ietf.org>
Subject: Re: [AVT] possible error on testvectors of draft "draft-ietf-avt-
	srtp-08.txt"
Date: Wed, 13 Aug 2003 10:51:38 +0200
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Hello,

As Claudio Thomas points out, there is an error in the test-vectors for
f8-mode present in the draft. To aid implementors of f8-mode we would
like to make more test-vectors available. It would be nice if anyone
with an implementation could verify the test-vectors retrievable
through this link: http://standards.ericsson.net/fli/drafts.html 
Our intention is to incorporate a reference to these in the next version.

Regards,
Karl



> Actual link: <http://www.ietf.org/internet-drafts/draft-ietf-avt-srtp-08.txt> >http://www.ietf.org/internet-drafts/draft-ietf-avt-srtp-08.txt
> 
> 
> Chapter B.1 AES-f8 Test Vectors (page 52)
> 
> 
> Position: j=2
> 
> 
> IV' XOR j : 595b699bbd3bc0df26062093c1ad8f70
> 
> Should be
> 
> IV' XOR j : 595b699bbd3bc0df26062093c1ad8f71
> 
> 
> Reason:
> 
> Last byte of IV' = 73, j=02: 73 xor 02 = 71 (0011 xor 0010 = 0001)
> 
> 
> If my calculator is right, then the values after this position also can't be correct.
> 
> 
> Do someone have already implemented it to give me working vextors? Or can somebody explain me >what is wrong on my calc?
> 
> 
> Thanks,
> 
> 
> Claudio


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Fri Aug 15 20:21:31 2003
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From: "Raymond Chen" <rchen@broadcom.com>
To: avt@ietf.org
Subject: RE: [AVT] Comments on draft-chen-rtp-bv-00.txt
Date: Fri, 15 Aug 2003 17:21:48 -0700
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Dear Magnus and Colin,

Thank you very much for taking the time to read our draft and for
providing useful comments.  We are reviewing your comments and all other
feedback and are working on a draft 01 to incorporate changes.  We hope
to have a revised version before long.

Thanks.

Raymond


-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com] 
Sent: Friday, July 25, 2003 8:28 AM
To: IETF AVT WG
Subject: [AVT] Comments on draft-chen-rtp-bv-00.txt


Hi,

Here are some comments on this draft. Before this I would like to say 
thank you to the authors for providing a very good initial version of 
the draft. I have very few comments.

1. There is need to tighten up the use of aggregation of consecutive 
frames. I would propose that in section 3.2 and 4.2 add a sentence 
stating that any frames in payload MUST be consecutive. If that is not 
intended there is need for having some mechanism for recovering the 
timestamp for each individual frame.

2. Section 3 and 4 are both missing specification for how to use the 
marker bit. I would suggest the normal definition that for packets with 
there first frame being an speech onset the marker bit is set to 1, 
otherwise 0. Please see draft-ietf-avt-profile-new-13.txt

3. I would also like to see some more text on how one recovers the RTP 
timestamp for an aggregated frame. I now that as long as you only intend

to aggregate consecutive frames it will be RTP TS + TS_rate*0.005*(n-1) 
however this should be written out.

4. I would also suggest that you tighten the language in Section 5 
regarding that BroadVoice 16 and 32 frames shall not be mixed in a file.

Please use the suitable RFC 2119 language for this.

5. Section 6.1, the usage of maxptime. Are there any need to define a 
default value for maxptime and especially of selecting 200 ms?

6. Section 6.1, Public Specification. This needs to be a complete 
reference.

7. Section 7. I would suggest that you improve your text for how to map 
the parameters to SDP. I would suggest borrowing text from section 8.3 
of RFC 3267.

8. The reference list needs to be split into normative and informative 
references.


Best Regards


Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt





_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Need the brief IETF copyright declaration at the beginning?

Abstract: isn't this intended to replace rather than merely update RFC 2833?

Page 2 first para: suggest "and" instead of "as well as" to provide more balance in 
the list of applications in the first sentence, and "In the second application.." 
rather than "Secondly.." further down.  Otherwise the reader is left searching for 
the second application.

Page 3 second-last para, last sentence, might be rephrased: "Busy tone is generated 
by the ISDN terminal or, for analog instruments, by the caller's switch, triggered 
by the appropriate ISUP message."

Page 4 first para might better read: "If a gateway cannot provide a tone 
representation, it SHOULD send the named signals and also send the audio tones as 
regular RTP audio packets ...".

Page 4, second-last para seems to have been cut off after "RFC 2198 [4] is used to".

Page 4 last para, first sentence: suggest replacing "to generate tone signals again" 
with "to regenerate the tones from the named events".  In the second sentence, add 
"their peer" in front of "Internet end systems".

Section 3.2 "Events and audio" bullet: missing "source" before "sends".

Last sentence of section 3.2: suggest replacing "in general should not interfere" by 
"are unlikely to interfere".

Section 3.3, compliance (immediately after the bullets).
First point: I'd suggest a subheading at this point: "3.3.1 Implementation 
considerations".

First para after bullets: to make things easier for the reader, add indication that 
Table 2 is in section 3.10.  To make it even easier, indicate that these are all 
DTMF events.  Suggested phrasing:
"A compliant implementation MUST support the DTMF events listed in Table 2 (section 
3.10)."
I don't mention the "Flash" exception because I don't think "Flash" belongs in 
section 3.10/Table 2.  (Further discussion below.)

Third para after bullets: change "render all tones in Table 6 the same ..." to 
"render all the country-specific line events in Table 6 (section 3.13) using the 
same tone ...".

Final para before 3.4: move up to immediately follow the bullets, particularly if 
the suggestion to add sub-heading 3.3.1 is accepted.

As a further improvement, add a final para to the main 3.3 (before the 3.3.1 
sub-heading), defining state transition events and describing the meaning of the 
"State?" column in the tables.  The text for this latter should be moved here from 
section 3.4, "Duration", where people can easily miss it.  Here's a suggestion:

"Certain events represent transitions into particular states.  They are indicated by 
the presence of "Yes" values in the column headed "State?" in the event tables 2 
through 6.  Notes to the tables indicate which states are mutually exclusive (that 
is, which states are terminated when a given state is reported).  Notes may also 
indicate other state-related restrictions on what events can be reported."

You can keep your current state number idea instead, but I personally feel that 
notes put the restrictions across more straightforwardly.

Section 3.4, "Timestamp": the statement
"The receiver calculates jitter for RTCP receiver reports based on all packets with 
a given timestamp."
is presumably a note for information rather than normative text.  Hence it seems 
appropriate to put "Note: " in front of this statement.  The sentence following this 
one seems to be completely off topic and could be omitted.

Section 3.4, "Duration":  as suggested above, move the general description of state 
transition events to 3.3.  Thus the second para becomes:
"The special duration value of zero is reserved to indicate that the event lasts 
"forever", i.e., is a state and is considered to be effective until updated. Only 
events marked with a "Yes" in the "state?" column in the tables below are allowed to 
use a duration value of zero. Other named events with such a duration SHOULD be 
ignored."

Soft-state: should we indicate any action for the receiver?  I see two possibilities:
- the receiver SHOULD consider the state "unknown" once it has expired without 
refresh, or
- the receiver MAY assume persistence of the state even if it has not been refreshed 
within the specified duration.

Section 3.4, E bit: the third para is mostly redundant.  Why not just say: "For 
named RTP events labeled "state" in Tables 2 through 6, sending of a packet with the 
"E" bit set is OPTIONAL."

I think the paragraphs following this one, beginning with "Receiver implementations 
MAY use different algorithms ..." and ending just before the R-bit description, 
might logically constitute a new subsection: "3.4.1 Use of payload data to generate 
tones".  This section would follow the R-bit description.

Within those paras, para beginning "Alternatively, ...": rephrase slightly to make 
clear the three stopping conditions for a tone:
"Alternatively, the receiver can start a tone and play it until
- it receives a packet with the "E" bit set, or
- it receives the next tone, indicated by a new timestamp value, or
- a given amount of time passes.
This is more robust ..."

Final para of that set, beginning "If a receiver has extended a tone ...": I suggest 
changing "when" to "if" and "that event" to "the same event instance".  Thus the 
sentence would read (with some additional words at the end):

"If a receiver has extended a tone by the maximum extension duration and started 
playing silence, it MUST NOT resume playing the tone if later packets arrive for the 
same event instance, as this would cause spurious events to be detected downstream."


Section 3.6, second para, last line has garbled punctuation -- or is it just that 
"in order to" begins a new sentence and a capital is needed?

If I understand that quotation from Q.24 (can't bring it up at the moment to check 
it), the part about "signalling velocity" means that the minimum interval from the 
start of one tone signal to the start of the next should be 93 ms.  Perhaps this is 
how it should be stated?

More from tomorrow's campsite.

Tom Taylor


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Greetings. Please find review comments for 2833bis-03 below (some more
nit-picking than others).

Section 2, 7th and 8th paragraph, Section 3.2 first half.
- The discussion in Section 2 about what representation to send seems to
ignore any negotiation that may have taken place. It is also somewhat
contrary to the first part of Section 3.2, which says that the source is free
to choose any of the four approaches. I think a bigger issue here is that the
document does not have an offer/answer section.

Section 3, 4th paragraph
- Last sentence says "RFC 2198 [4] is used to". Either something is missing
here or this should be deleted.

Section 3.2, first list item (Events and audio)
- It is not clear to me whether redundancy encoding (RFC 2198) is used here
or not and it should probably be spelled out.

Section 3.2, first item in list
OLD: PCMU or the codec
NEW: PCMU or another codec
             ^^^^^^^^

Section 3.3, second paragraph
- Text currently says "MUST support the events listed in Table 2 with the
exception of flash". We discussed this for 2833 in the past, and agreed that
(in practice) only the events you actually declare support for are supported
(there was an implementation note to that effect at
http://www.cs.columbia.edu/~hgs/rtp/payload.html in the past, but this URL no
longer works). In other words, you don't necessarily have to support all DTMF
tones to support RFC2833bis.

Section 3.3, second last paragraph:
- There is no normative language in this section, but nevertheless, it would
probably be worthwhile clarifying that this has no bearing on implementations
(in fact, I don't agree completely with the statements in this paragraph and
would consider removing it).

Section 3.5, duration field bullet
- Text says "all but the last event have the maximum duration expressible in
the duration field". Should then explain what the last event actually has in
this field.

Section 3.5, E field bullet, second paragraph:
- Text says "until retransmitting the last packet for a tone". There should
either be a forward reference to Section 3.6 to explain what "retransmit"
means (since it's not just a single packet retransmission, but rather a
resend of the event with a new sequence number or use of 2198) or it should
be explained here.

Section 3.5, E field bullet, third paragraph:
- Text says "the appearance of a different named event implies the end of the
previous state". That's not entirely accurate; only if the new event belongs
to the same "state class" as the old state will the state be changed (e.g.
receipt of a DTMF digit does not end the off-hook state).

Section 3.5, E field bullet, second last paragraph:
- Text says "A slight extension of tone durations and shortening of pauses is
generally harmless". "Generally" is the keyword here, as there are cases
where it is not harmless and implementations need to do their absolute best
to maintain the original timing - this may be worth pointing out.

Section 3.6, 4th paragraph:
- Text says "If an event has ended and there bas been no new event in the
last interval, the event with final duration SHOULD be sent a total of three
times at the interval used by the source for the updates". When subevents had
been sent, the duration will not really be the "final duration". Also, let's
say a new event occurs after I sent the final event two times; do I still
send the old event a third time ?

Section 3.6, 4th paragraph, last sentence:
- Text says "The last update is also repeated at the end of subevents, as
there is no other way to detect the duration of the event". Does this also
apply if a new event happens right after the end of the subevent ? Either
way, it's probably worth clarifying.

Section 3.9:
- There should be offer/answer considerations here (probably a separate
sub-section). In particular, how the "fmtp" parameter is processed should be
discussed (e.g. seems reasonable that, within the context of offer/answer, it
is actually negotiated). Also, for sendonly or sendrecv streams, it seems
reasonable the events can not only be received, but also sent.

Section 3.10, second paragraph:
- Text says "The Flash event must only be sent when the state is off hook".
What does "state" refer to here (e.g. do we have to have an off-hook state
signaled via the 2833bis off-hook event in order to use this) ? Also, "must"
should probably be "MUST".

Section 3.11, /ANSam event, third paragraph
- Let's say we send ANS, get a phase reversal, and then send /ANS. The text
is not entirely clear on what happens from here. If the expected phase
reversals continue to occur, I believe we continue sending the /ANS which
should probably be stated. Also, in that case, is it still one and the same
event, and hence the timestamp continues unchanged ? If I don't get the
expected phase reversal, do I then send ANS again ? Clarification on these
points would be good.

Section 3.11, V8bis signals
- I know little about V8bis and could easily be wrong here, but some of the
V8bis signals look suspicious to me, so unless somebody that knows about this
has reviewed it, I'd be concerned. It seems that only a subset of the V8bis
signals are included and that you would need more to implement some of the
V8bis state transitions (see e.g. Figure 11 in V8.bis). Also, based on a
cursory look at V8bis, the definition of  CRDi does not seem correct (if I
understand correctly, CRDi would not be issued in response to CRe; CRDr
would). Tones affected by this comment are: CRDi, CRDr, CRe, ESi, ESr, MRdi,
MRdr, MRe.

Section 3.11, CI:
- Text says "To fully express the call indicator, it would be followed by a
call function octet, composed of individual V.21 bit events". It's unclear if
the expectation here is to include the stop and start bit as well (producing
10 bits) or whether only the call function octet itself is expected to be
sent, in which case the start and stop bits will have to be generated by the
receiver if the tone is to be sent out.

Section 3.11, Table 3:
- For the "V.25, echo canceller disable" row, it's not clear whether the
"ANS, /ANS, ANS, /ANS" indication refers to the ITU-T spec or the 2833bis. I
assume it's the former, since the representation in 2833bis would rather be
"ANS, /ANS........" (I believe). If so, this should probably be clarified.

Section 3.12, 3.13, and 5:
- The example in Section 5 surprised me a little inasmuch as it included
silence as part of the tone. Thinking further about this, it leads to a more
general question about how some of the "tones" in Section 3.12 and 3.13 (and
maybe others, e.g. seems similar to the /ANS, /ANSam question), which are
actually not just tones, but tone patterns, are represented and rendered. In
the case of ringing, is each "event" a single ring, a single ring-cycle, or
does it continue ? What about trailing silence in a tone pattern in general ?
For example, confirmation tone as defined in GR-506-CORE ends with a steady
application of tone off, so when does it end ? I think we need more
description of these issues in general in the definition of these kinds of
tones.

Section 3.13:
- At a minimum, we need to have references for the tones in this section (and
preferably some text too). Right now these event numbers could be anything.

Section 3.14, ABCD transitional and Trunk Unavailable
- Text says "At the time of a transition, the same ABCD information is sent 3
times at an interval of 5 ms". Seems a bit strange to me that the general
retransmission strategy is not being used here.  What is the reasoning behind
this ? A similar comment applies to the "Trunk unavailable" event.

Section 3.14, Continuity Tones:
- Both the 1780 and the 2010 can be either sent or received (depending on the
switch configuration at either end).

Section 3.14:
- I have asked another person to review this section as well, so there may be
additional comments here.

Section 4.3:
- May be worth describing how the Marker bit is used.

Section 7, and References
- Reference to RFC 1889 and RFC 1890 should be replaced with RFC 3550 and RFC
3551.

Normative References:
- [3]: Latest version of V.8 is 11/00
- [9]: Latest version of T.30 is 07/03
- [11]: Latest version of V.8bis is 11/00
- [16]: Latest version of Q.440 is 11/88 (not 11/98)
- [17]: Latest version of Q.140 is 11/88 (not 11/98)
- [18]: Latest version of Q.151 is 11/88 (not 11/98)
- [19]: Latest version of I.366.2 is 11/00
- [20]: Latest version of Q.400 is 11/88 (not 11/98)
- [21]: Latest version of Q.411 is 11/88 (not 11/98)
- [22]: Latest version of Q.421 is 11/88 (not 11/98)
- [24]: Latest version of E.180 is 03/98. Also delete "Recommendation
Supplement 2 to" from the text.

In addition to the above, I have some strictly editorial comments which I
will send separately to just the author.

-- Flemming

Colin Perkins wrote:

> On Monday, Jul 28, 2003, at 02:15 Europe/London, Colin Perkins wrote:
> > This is to announce a working group last call on the RTP Payload for
> > DTMF
> > Digits, Telephony Tones and Signals <draft-ietf-avt-rfc2833bis-03.txt>.
> > Please send any final comments on this draft to the mailing list by
> > 11th
> > August 2003. If no substantive technical issues are raised, we intend
> > to
> > submit this draft for publication as a proposed standard RFC to
> > obsolete
> > RFC 2833.
> >
> > Note that this draft will not be advanced until we receive at least one
> > valid review from someone not directly involved with production of the
> > draft (i.e., not an author or contributor).
>
> This is to solicit reviews of the draft-ietf-avt-rfc2833bis-03.txt. As
> noted above, we will not advance the draft until at least one valid
> technical review, from someone not directly involved with the
> production of the draft, is received.
>
> --
> Colin Perkins
> http://csperkins.org/
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt


_______________________________________________
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https://www1.ietf.org/mailman/listinfo/avt



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From: "Oren Peleg" <OrenP@audiocodes.com>
To: <avt@ietf.org>
Subject: [AVT] RTP/RTCP Simulator
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Hi all,

I am searching for an RTP/RTCP simulator for RTP/RTCP protocol testing. =
Should include advance RTP/RTCP control capabilities (E.g. working with =
Padding, extensions, etc.). Can anyone recommend on such a product?

Thanks,

Oren Peleg

Work phone No. : 03 - 5394167
Home phone No. : 03 - 5335913
Cell phone No. : 055 - 669613
Email : Oren.Peleg@audiocodes.com


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<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">Hi all,</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">I am searching for an =
RTP/RTCP simulator for RTP/RTCP protocol testing. Should include advance =
RTP/RTCP control capabilities (E.g. working with Padding, extensions, =
etc.). Can anyone recommend on such a product?</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">Thanks,</FONT></P>

<P DIR=3DLTR><FONT COLOR=3D"#000080" SIZE=3D4 FACE=3D"Monotype =
Corsiva">Oren Peleg</FONT></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Book Antiqua">Work phone No. : 03 - 5394167</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Book Antiqua">Home phone No. : 03 - 5335913</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Book Antiqua">Cell phone No. : 055 - 669613</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Book =
Antiqua">Email :<U> </U></FONT><U><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Book =
Antiqua">Oren.Peleg@audiocodes.com</FONT></U></SPAN><U><SPAN =
LANG=3D"he"></SPAN></U><SPAN LANG=3D"he"></SPAN></P>

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_______________________________________________
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avt@ietf.org
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From exim@www1.ietf.org  Wed Aug 20 08:35:30 2003
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From: "Jose Rey" <rey@panasonic.de>
To: "Colin Perkins" <csp@csperkins.org>
Cc: <avt@ietf.org>
Subject: RE: [AVT] 3GPP timed text draft: more generic?
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Hi Colin, all,


> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]
> Sent: Monday, August 11, 2003 12:03 PM
> To: Jose Rey
> Cc: avt@ietf.org
> Subject: Re: [AVT] 3GPP timed text draft: more generic?
>
>
> On Monday, Aug 11, 2003, at 09:15 Europe/London, Jose Rey wrote:
> > Colin Perkins wrote:
> >> A specific payload format for 3GPP timed text might be
> >> appropriate, but I
> >> would like to understand the options and alternatives before
> >> we make that
> >> choice. Is 3GPP the only body doing timed and/or decorated
> >> text? If not,
> >> what are the alternatives, and how do they relate?
> >>
> >> --
> >
> > at the moment there are at least two more standardisation bodies
> > interested in timed text: MPEG and W3C.
> >
> > - MPEG is currently defining their own TT format based on
> 3GPP TT (see
> > Committee Draft of ISO/IEC 14496-17 from July'03 Meeting in
> Trondheim)
> > by defining the access unit structure, the stream type, the
> object type
> > indication and decoder specific info of the timed text samples. The
> > definition of these items enables the transport over RTP using the
> > mpeg4simple draft. The work in MPEG is assumed to be
> finalised around
> > July 2004.
> >
> > - W3C is at a very early stage in defining a timed text authoring
> > format. It has recently (May'03) published a draft for use
> cases and
> > requirements of a timed text authoring format:
> > http://www.w3.org/TR/2003/WD-tt-af-1-0-req-20030515/. This
> draft has a
> > rather long list of requirements and lists many
> functionalities. In my
> > understanding W3C targets to a generic format with rich
> functionality.
> > I
> > don't know how quick it may develop (anyone?).
>
> I believe SMPTE may also have timed text work in their Digital Cinema
> group?
>

Yes, there is a DC28.2W Subtitles and Captions Adhoc Group in SMPTE
dealing with a stream timed text format. I've checked the website and as
far as I understand from the available documents they intend to develop
a media format to stream text encoded as MNG (Multi-image Network
Graphics) images. On April'02 they were still working on a requirements
document for the stream text format but no output has been published
until today, at least publicly.

Question to the group: is anyone aware of the progress of this item?

thanks,

Jose



> > None of these above is an option for us, because 1) the
> only timed text
> > format in 3GPP is 3GPP timed text 2) 3GPP has not adopted the
> > mpeg4simple for
> > transport and 3) for this 3GPP timed text format we need an
> RTP payload
> > format within the current 3GPP Release 6 schedule (i.e. March'04).
>
> I agree that using the MPEG4-simple format doesn't make
> sense, and that
> the W3C schedule looks to be too long for 3GPP requirements.
>
> --
> Colin Perkins
> http://csperkins.org/
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Alan Clark" <alan@telchemy.com>
To: "'Oren Peleg'" <OrenP@audiocodes.com>, <avt@ietf.org>
Subject: RE: [AVT] RTP/RTCP Simulator
Date: Wed, 20 Aug 2003 09:46:57 -0400
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RTP/RTCP SimulatorOren

We have software for both RTP/ RTCP generation and analysis, RTCP XR
generation and parsing and also for trace file generation and analysis.

Regards

Alan Clark
Telchemy


-----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Oren Peleg
Sent: Wednesday, August 20, 2003 8:13 AM
To: avt@ietf.org
Subject: [AVT] RTP/RTCP Simulator


Hi all,

I am searching for an RTP/RTCP simulator for RTP/RTCP protocol testing.
Should include advance RTP/RTCP control capabilities (E.g. working with
Padding, extensions, etc.). Can anyone recommend on such a product?

Thanks,

Oren Peleg

Work phone No. : 03 - 5394167

Home phone No. : 03 - 5335913

Cell phone No. : 055 - 669613

Email : Oren.Peleg@audiocodes.com

------=_NextPart_000_0040_01C366FF.FF8955F0
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<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dwindows-1255">
<TITLE>RTP/RTCP Simulator</TITLE>

<META content=3D"MSHTML 6.00.2800.1106" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D762324413-20082003><FONT color=3D#0000ff=20
size=3D2>Oren</FONT></SPAN></DIV>
<DIV><SPAN class=3D762324413-20082003><FONT color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D762324413-20082003><FONT color=3D#0000ff size=3D2>We =
have software=20
for both RTP/ RTCP generation and analysis, RTCP XR generation and =
parsing and=20
also for trace file generation and analysis.&nbsp;</FONT></SPAN></DIV>
<DIV><SPAN class=3D762324413-20082003><FONT color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D762324413-20082003><FONT color=3D#0000ff=20
size=3D2>Regards</FONT></SPAN></DIV>
<DIV><SPAN class=3D762324413-20082003><FONT color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D762324413-20082003><FONT color=3D#0000ff =
size=3D2>Alan=20
Clark</FONT></SPAN></DIV>
<DIV><SPAN class=3D762324413-20082003><FONT color=3D#0000ff=20
size=3D2>Telchemy</FONT></SPAN></DIV>
<DIV><SPAN class=3D762324413-20082003><FONT color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D762324413-20082003><FONT color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
size=3D2>-----Original Message-----<BR><B>From:</B> avt-admin@ietf.org=20
[mailto:avt-admin@ietf.org]<B>On Behalf Of </B>Oren =
Peleg<BR><B>Sent:</B>=20
Wednesday, August 20, 2003 8:13 AM<BR><B>To:</B> =
avt@ietf.org<BR><B>Subject:</B>=20
[AVT] RTP/RTCP Simulator<BR><BR></FONT></DIV><!-- Converted from =
text/rtf format -->
<P dir=3Dltr><FONT face=3DArial size=3D2>Hi all,</FONT></P>
<P dir=3Dltr><FONT face=3DArial size=3D2>I am searching for an RTP/RTCP =
simulator for=20
RTP/RTCP protocol testing. Should include advance RTP/RTCP control =
capabilities=20
(E.g. working with Padding, extensions, etc.). Can anyone recommend on =
such a=20
product?</FONT></P>
<P dir=3Dltr><FONT face=3DArial size=3D2>Thanks,</FONT></P>
<P dir=3Dltr><FONT face=3D"Monotype Corsiva" color=3D#000080 =
size=3D4>Oren=20
Peleg</FONT></P>
<P dir=3Dltr><SPAN lang=3Den-us><FONT face=3D"Book Antiqua" =
color=3D#000000 size=3D2>Work=20
phone No. : 03 - 5394167</FONT></SPAN></P>
<P dir=3Dltr><SPAN lang=3Den-us><FONT face=3D"Book Antiqua" =
color=3D#000000 size=3D2>Home=20
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color=3D#000000 size=3D2>Cell=20
phone No. : 055 - 669613</FONT></SPAN></P>
<P dir=3Dltr><SPAN lang=3Den-us><FONT face=3D"Book Antiqua" =
size=3D2>Email :<U>=20
</U></FONT><U><FONT face=3D"Book Antiqua" color=3D#0000ff=20
size=3D2>Oren.Peleg@audiocodes.com</FONT></U></SPAN><U><SPAN=20
lang=3Dhe></SPAN></U><SPAN lang=3Dhe></SPAN></P></BODY></HTML>

------=_NextPart_000_0040_01C366FF.FF8955F0--


_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hi,
    I am facing an issue in inter-stream (audio-video) synchronization 
between two RTP streams. The NTP-RTP timestamp correlation comes in the 
RTCP packets, but the RFC does not specify the units in which the NTP is 
to be specified. This causes a problem in calculating the audio-video 
synchronization.
    As far as I understand, the Synchronization is not possible unless 
the unit of the NTP is known exactly, which can only be known after the 
second RTCP packet arrives (from which you can indirectly judje the NTP 
granularity and hence it's unit). Are there any known solutions to this 
problem ?
Thanks
Avinash


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Aug 20 12:12:28 2003
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On Thursday, Aug 21, 2003, at 02:42 Europe/London, Avinash Bhatia wrote:
>    I am facing an issue in inter-stream (audio-video) synchronization 
> between two RTP streams. The NTP-RTP timestamp correlation comes in 
> the RTCP packets, but the RFC does not specify the units in which the 
> NTP is to be specified. This causes a problem in calculating the 
> audio-video synchronization.
>    As far as I understand, the Synchronization is not possible unless 
> the unit of the NTP is known exactly, which can only be known after 
> the second RTCP packet arrives (from which you can indirectly judje 
> the NTP granularity and hence it's unit). Are there any known 
> solutions to this problem ?

RTP makes no guarantee of the accuracy or resolution of the clock the 
sender is using to generate the timestamp. However, the format of the 
timestamp is precisely specified in Section 4 of RFC 3550 (the upper 32 
bits are seconds, the lower 32 bits fractions of a second, since 1 
January 1900).

-- 
Colin Perkins
http://csperkins.org/


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Aug 20 13:38:32 2003
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From: "Allison, Art" <AAllison@nab.org>
To: "'Colin Perkins'" <csp@csperkins.org>,
        Avinash Bhatia
	 <abhatia@neomagic.com>
Cc: avt@ietf.org, Vivek Mudgil <vmudgil@neomagic.com>
Subject: RE: [AVT] RTCP timestamp units
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Colin
By saying (no guarantee of ... resolution) it seems you are saying that the
RFC does not actually require all 32 bits to be incremented. If so one can
chose how many bits to send with real values, and zero fill the rest.
 Is that correct?
I would think that the resolution would be the precision of the number, and
the value of the lsb would establish absolutely the resolution (but it may
be offset by an unknown amount from real time which offset could vary - i.e.
no accuracy).

Art
::{)
Art Allison
Director Advanced Engineering
NAB
1771 N St NW
Washington DC 20036
202 429 5418


-----Original Message-----
From: Colin Perkins [mailto:csp@csperkins.org]
Sent: Wednesday, August 20, 2003 12:11 PM
To: Avinash Bhatia
Cc: avt@ietf.org; Vivek Mudgil
Subject: Re: [AVT] RTCP timestamp units


On Thursday, Aug 21, 2003, at 02:42 Europe/London, Avinash Bhatia wrote:
>    I am facing an issue in inter-stream (audio-video) synchronization 
> between two RTP streams. The NTP-RTP timestamp correlation comes in 
> the RTCP packets, but the RFC does not specify the units in which the 
> NTP is to be specified. This causes a problem in calculating the 
> audio-video synchronization.
>    As far as I understand, the Synchronization is not possible unless 
> the unit of the NTP is known exactly, which can only be known after 
> the second RTCP packet arrives (from which you can indirectly judje 
> the NTP granularity and hence it's unit). Are there any known 
> solutions to this problem ?

RTP makes no guarantee of the accuracy or resolution of the clock the 
sender is using to generate the timestamp. However, the format of the 
timestamp is precisely specified in Section 4 of RFC 3550 (the upper 32 
bits are seconds, the lower 32 bits fractions of a second, since 1 
January 1900).

-- 
Colin Perkins
http://csperkins.org/


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Aug 20 14:18:28 2003
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        Vivek Mudgil <vmudgil@neomagic.com>
To: "Allison, Art" <AAllison@nab.org>
From: Colin Perkins <csp@csperkins.org>
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Art,

On Wednesday, Aug 20, 2003, at 18:37 Europe/London, Allison, Art wrote:
> By saying (no guarantee of ... resolution) it seems you are saying 
> that the
> RFC does not actually require all 32 bits to be incremented. If so one 
> can
> chose how many bits to send with real values, and zero fill the rest.
>  Is that correct?

That's one possible implementation choice for generating the NTP-format 
timestamp in RTCP SR packets, yes. However, all 32 bits of the RTP 
timestamp MUST be incremented.

> I would think that the resolution would be the precision of the 
> number, and
> the value of the lsb would establish absolutely the resolution (but it 
> may
> be offset by an unknown amount from real time which offset could vary 
> - i.e.
> no accuracy).

The sender may not know the resolution of its clock, and hence the LSBs 
of the NTP format timestamp may be garbage. I expect this will make it 
difficult to infer the characteristics of the clock from the timestamp 
values.

Of course, if RTP is used in a controlled environment, it may be 
possible to mandate that senders use a clock with tightly-specified 
characteristics in that environment.

-- 
Colin Perkins
http://csperkins.org/


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] On draft-hellstrom-avt-rfc2793bis-00.txt
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Folks,

I wanted to call your attention to draft-hellstrom-avt-rfc2793bis-00.txt,
which was just made available as an Internet Draft.

There are a number of enhancements, primarily focused on allowing text and
audio transmitted between two trunking gateways to be switched via a payload
type.  Other minor modifications were made, including the recommendation of
using 3 levels of redundancy to reduce the probability of losing any
transmitted
text and language to improve how packets get (re)transmitted.  The document
also proposes the registration of an audio/t140 MIME type for the benefit of
GWs that will be transmitting received audio and text to another device over
IP.

I would like to see this accepted as a work item for the AVT working group.
ITU-T SG16 is working to produce a standard for transport of PSTN textphone
signals over an IP network between two trunking gateways.  This work will be
vitally important for the completion of that standard, which they hope to
approve in January 2004.  (Yes, it's an aggressive schedule.)

Already, I've received a comment suggesting that fmtp should be used to
allow a device to specify a "character per second" transmission rate.  I'll
revise the next revision accordingly, but I'd certainly like to hear any
other comments on this draft.

Thanks!
Paul


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Vivek Mudgil <vmudgil@neomagic.com>
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To: Colin Perkins <csp@csperkins.org>
CC: "Allison, Art" <AAllison@nab.org>, Avinash Bhatia
 <abhatia@neomagic.com>,
        avt@ietf.org
Subject: Re: [AVT] RTCP timestamp units
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Section 4 of RFC 3550 clearly specifies that for RTP purposes, only 
differences between pairs of NTP timestamps are used. This implies that
for inter-media synchronisation, the receiver should have received at
least 2 RTCP SR reports. Which in turn imply that inter-media 
synchronisation in playback is only possible after a minimum of 5 sec 
(min RTCP Interval). Is this implication right or am I missing something?

Even if first NTP timestamp is used for synchronisation,, assuming that
the NTP time stamp is provided as specified in Section 4, how will a
receiver synchronise the two media till the first SR report is received?

Colin Perkins wrote:
> Art,
> 
> On Wednesday, Aug 20, 2003, at 18:37 Europe/London, Allison, Art wrote:
> 
>> By saying (no guarantee of ... resolution) it seems you are saying 
>> that the
>> RFC does not actually require all 32 bits to be incremented. If so one 
>> can
>> chose how many bits to send with real values, and zero fill the rest.
>>  Is that correct?
> 
> 
> That's one possible implementation choice for generating the NTP-format 
> timestamp in RTCP SR packets, yes. However, all 32 bits of the RTP 
> timestamp MUST be incremented.
> 
>> I would think that the resolution would be the precision of the 
>> number, and
>> the value of the lsb would establish absolutely the resolution (but it 
>> may
>> be offset by an unknown amount from real time which offset could vary 
>> - i.e.
>> no accuracy).
> 
> 
> The sender may not know the resolution of its clock, and hence the LSBs 
> of the NTP format timestamp may be garbage. I expect this will make it 
> difficult to infer the characteristics of the clock from the timestamp 
> values.
> 
> Of course, if RTP is used in a controlled environment, it may be 
> possible to mandate that senders use a clock with tightly-specified 
> characteristics in that environment.
> 




_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: Re: [AVT] RTCP timestamp units
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The first query is valid in context of Section 6.3.1 of RFC 1889, but 
does not hold any more (i.e., since 3550), since the line "It is 
permissible to use the sampling clock to estimate elapsed wallclock 
time." has been removed.
However, it still says that "A sender that has no notion of wallclock or 
elapsed time may set the NTP timestamp to zero.". In such a case, how is 
the reciever of multiple streams expected to do inter-stream 
synchronization ?

Vivek Mudgil wrote:

> Section 4 of RFC 3550 clearly specifies that for RTP purposes, only 
> differences between pairs of NTP timestamps are used. This implies that
> for inter-media synchronisation, the receiver should have received at
> least 2 RTCP SR reports. Which in turn imply that inter-media 
> synchronisation in playback is only possible after a minimum of 5 sec 
> (min RTCP Interval). Is this implication right or am I missing something?
>
> Even if first NTP timestamp is used for synchronisation,, assuming that
> the NTP time stamp is provided as specified in Section 4, how will a
> receiver synchronise the two media till the first SR report is received?
>
> Colin Perkins wrote:
>
>> Art,
>>
>> On Wednesday, Aug 20, 2003, at 18:37 Europe/London, Allison, Art wrote:
>>
>>> By saying (no guarantee of ... resolution) it seems you are saying 
>>> that the
>>> RFC does not actually require all 32 bits to be incremented. If so 
>>> one can
>>> chose how many bits to send with real values, and zero fill the rest.
>>>  Is that correct?
>>
>>
>>
>> That's one possible implementation choice for generating the 
>> NTP-format timestamp in RTCP SR packets, yes. However, all 32 bits of 
>> the RTP timestamp MUST be incremented.
>>
>>> I would think that the resolution would be the precision of the 
>>> number, and
>>> the value of the lsb would establish absolutely the resolution (but 
>>> it may
>>> be offset by an unknown amount from real time which offset could 
>>> vary - i.e.
>>> no accuracy).
>>
>>
>>
>> The sender may not know the resolution of its clock, and hence the 
>> LSBs of the NTP format timestamp may be garbage. I expect this will 
>> make it difficult to infer the characteristics of the clock from the 
>> timestamp values.
>>
>> Of course, if RTP is used in a controlled environment, it may be 
>> possible to mandate that senders use a clock with tightly-specified 
>> characteristics in that environment.
>>
>
>
>
>



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: Re: [AVT] RTCP timestamp units
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From: Colin Perkins <csp@csperkins.org>
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On Thursday, Aug 21, 2003, at 05:54 Europe/London, Vivek Mudgil wrote:
> Section 4 of RFC 3550 clearly specifies that for RTP purposes, only 
> differences between pairs of NTP timestamps are used. This implies > that
> for inter-media synchronisation, the receiver should have received at
> least 2 RTCP SR reports. Which in turn imply that inter-media 
> synchronisation in playback is only possible after a minimum of 5 sec 
> (min RTCP Interval). Is this implication right or am I missing 
> something?

In order to perform lip-sync you need to receive one RTCP SR packet 
relating to the audio stream, and one RTCP SR packet relating to the 
video stream. This pair of NTP-format timestamps, and their 
corresponding RTP timestamps, is used to align the media.
This means there is an initial delay - on average 2.5 seconds duration 
if you use the default RTCP timing rules - when the receiver does not 
have the necessary information to perform lip-sync.

> Even if first NTP timestamp is used for synchronisation,, assuming that
> the NTP time stamp is provided as specified in Section 4, how will a
> receiver synchronise the two media till the first SR report is 
> received?

The receiver cannot synchronise media streams until an RTCP SR packet 
has been received from each stream.

-- 
Colin Perkins
http://csperkins.org/


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: Re: [AVT] RTCP timestamp units
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On Thursday, Aug 21, 2003, at 18:29 Europe/London, Avinash Bhatia wrote:
> The first query is valid in context of Section 6.3.1 of RFC 1889, but 
> does not hold any more (i.e., since 3550), since the line "It is 
> permissible to use the sampling clock to estimate elapsed wallclock 
> time." has been removed.
> However, it still says that "A sender that has no notion of wallclock 
> or elapsed time may set the NTP timestamp to zero.". In such a case, 
> how is the reciever of multiple streams expected to do inter-stream 
> synchronization ?

If the sender sets the NTP timestamp to zero, it is not possible for a 
receiver to do inter-stream synchronisation.

-- 
Colin Perkins
http://csperkins.org/


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Thu, 21 Aug 2003 18:34:51 +0100
Subject: Re: [AVT] Re: I-D ACTION:draft-jones-avt-audio-t38-00.txt]
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From: Colin Perkins <csp@csperkins.org>
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Hi,

It seems that this draft needs more motivation and background 
information in the introduction, to make it clear exactly what is being 
registered and the scope of applicability for this MIME type. An 
additional section "Relation to RFC 3362" would also be appropriate, I 
believe.

I also have a number of other comments:

  - Bibliographic citations are not allowed in the abstract, you should 
move
    them into the introduction.
  - The "Conventions used in this document" section should be a numbered
    section, after the introduction.
  - The draft needs copyright and IPR notices adding
  - The current section 2 can probably be included under the "Encoding
    Considerations" header of the MIME registration.
  - Section 3 should be titled "IANA Considerations"
  - Section 3: The RTP timestamp rate is presumably a required parameter?
  - The draft needs a brief section explaining how the MIME name and any
    parameters are mapped onto SDP (e.g. see Section 5.1 of RFC 3557).
  - Security considerations needs to reference the security 
considerations
    of the RTP spec and any applicable profiles, along with any 
equivalent
    section in T.38 itself.
  - References need to be split into Normative and Informative sections.

You might also consider reading http://www.ietf.org/ID-nits.html where 
a lot of these issues are discussed.

Cheers,
Colin




On Wednesday, Aug 13, 2003, at 03:21 Europe/London, Paul E. Jones wrote:
> Tom,
>
> The purpose is to register the MIME type "audio/t38".  The reason is 
> that
> work is underway within ITU-T Q14/16 to support the carriage of T.38's 
> IFP
> packets through RTP.  This will allow us to then signal a single "m=" 
> like
> that looks like this:
>
>   m=audio 5600 RTP/AVP 18 98
>   a=rtpmap:98 t38/8000
>
> When two media gateways have negotiated this (or similar) SDP and a 
> fax tone
> is detected on the line side, a mode shift may occur by simply 
> transmitting
> packets with PT=98, rather than PT=18.  This reduces the need for 
> either
> consuming additional UDP ports per call for image/t38 or going through 
> a
> lengthy exchange of messages to open a fax channel.
>
> Paul
>
> ----- Original Message -----
> From: "Tom Taylor" <taylor@nortelnetworks.com>
> To: <avt@ietf.org>
> Cc: <paulej@packetizer.com>
> Sent: Tuesday, August 12, 2003 11:10 AM
> Subject: [Fwd: I-D ACTION:draft-jones-avt-audio-t38-00.txt]
>
>
>> What's this all about?  First point is that image/t38 is already a
> registered MIME
>> type.  Second point is that T.38 does NOT use RTP transport, in
> contradiction to the
>> statement in the I-D that it does.
>>
>> -------- Original Message --------
>> Subject: I-D ACTION:draft-jones-avt-audio-t38-00.txt
>> Date: Mon, 11 Aug 2003 15:46:16 -0400
>> From: Internet-Drafts@ietf.org
>> Reply-To: Internet-Drafts@ietf.org
>> To: IETF-Announce: ;
>>
>> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>>
>>
>> Title : Real-Time Facsimile (T.38) - audio/t38 MIME Sub-type
>>                            Registration
>> Author(s) : P. Jones
>> Filename : draft-jones-avt-audio-t38-00.txt
>> Pages : 4
>> Date : 2003-8-11
>>
>> This document defines the MIME sub-type audio/t38.  The packetization
>> and usage of this MIME type, which is intended for use within SDP [2]
>> and H.248 [3] systems, is specified within ITU-T Recommendation T.38
>> [1].
>>
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-jones-avt-audio-t38-00.txt
>>
>> To remove yourself from the IETF Announcement list, send a message to
>> ietf-announce-request with the word unsubscribe in the body of the
> message.
>>
>> Internet-Drafts are also available by anonymous FTP. Login with the
> username
>> "anonymous" and a password of your e-mail address. After logging in,
>> type "cd internet-drafts" and then
>> "get draft-jones-avt-audio-t38-00.txt".
>>
>> A list of Internet-Drafts directories can be found in
>> http://www.ietf.org/shadow.html
>> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>> ...
>>
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>
>
-- 
Colin Perkins
http://csperkins.org/


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Paul E. Jones" <paulej@packetizer.com>
To: "Colin Perkins" <csp@csperkins.org>
Cc: "Tom Taylor" <taylor@nortelnetworks.com>, <avt@ietf.org>
References: <C4EC0B2E-D3FD-11D7-8D51-000A957FC5F2@csperkins.org>
Subject: Re: [AVT] Re: I-D ACTION:draft-jones-avt-audio-t38-00.txt]
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Colin,

Thanks for the feedback.  I will try to incorporate your requests into a new
revision of the draft over the next few days.

Paul

----- Original Message ----- 
From: "Colin Perkins" <csp@csperkins.org>
To: "Paul E. Jones" <paulej@packetizer.com>
Cc: "Tom Taylor" <taylor@nortelnetworks.com>; <avt@ietf.org>
Sent: Thursday, August 21, 2003 1:34 PM
Subject: Re: [AVT] Re: I-D ACTION:draft-jones-avt-audio-t38-00.txt]


> Hi,
>
> It seems that this draft needs more motivation and background
> information in the introduction, to make it clear exactly what is being
> registered and the scope of applicability for this MIME type. An
> additional section "Relation to RFC 3362" would also be appropriate, I
> believe.
>
> I also have a number of other comments:
>
>   - Bibliographic citations are not allowed in the abstract, you should
> move
>     them into the introduction.
>   - The "Conventions used in this document" section should be a numbered
>     section, after the introduction.
>   - The draft needs copyright and IPR notices adding
>   - The current section 2 can probably be included under the "Encoding
>     Considerations" header of the MIME registration.
>   - Section 3 should be titled "IANA Considerations"
>   - Section 3: The RTP timestamp rate is presumably a required parameter?
>   - The draft needs a brief section explaining how the MIME name and any
>     parameters are mapped onto SDP (e.g. see Section 5.1 of RFC 3557).
>   - Security considerations needs to reference the security
> considerations
>     of the RTP spec and any applicable profiles, along with any
> equivalent
>     section in T.38 itself.
>   - References need to be split into Normative and Informative sections.
>
> You might also consider reading http://www.ietf.org/ID-nits.html where
> a lot of these issues are discussed.
>
> Cheers,
> Colin
>
>
>
>
> On Wednesday, Aug 13, 2003, at 03:21 Europe/London, Paul E. Jones wrote:
> > Tom,
> >
> > The purpose is to register the MIME type "audio/t38".  The reason is
> > that
> > work is underway within ITU-T Q14/16 to support the carriage of T.38's
> > IFP
> > packets through RTP.  This will allow us to then signal a single "m="
> > like
> > that looks like this:
> >
> >   m=audio 5600 RTP/AVP 18 98
> >   a=rtpmap:98 t38/8000
> >
> > When two media gateways have negotiated this (or similar) SDP and a
> > fax tone
> > is detected on the line side, a mode shift may occur by simply
> > transmitting
> > packets with PT=98, rather than PT=18.  This reduces the need for
> > either
> > consuming additional UDP ports per call for image/t38 or going through
> > a
> > lengthy exchange of messages to open a fax channel.
> >
> > Paul
> >
> > ----- Original Message -----
> > From: "Tom Taylor" <taylor@nortelnetworks.com>
> > To: <avt@ietf.org>
> > Cc: <paulej@packetizer.com>
> > Sent: Tuesday, August 12, 2003 11:10 AM
> > Subject: [Fwd: I-D ACTION:draft-jones-avt-audio-t38-00.txt]
> >
> >
> >> What's this all about?  First point is that image/t38 is already a
> > registered MIME
> >> type.  Second point is that T.38 does NOT use RTP transport, in
> > contradiction to the
> >> statement in the I-D that it does.
> >>
> >> -------- Original Message --------
> >> Subject: I-D ACTION:draft-jones-avt-audio-t38-00.txt
> >> Date: Mon, 11 Aug 2003 15:46:16 -0400
> >> From: Internet-Drafts@ietf.org
> >> Reply-To: Internet-Drafts@ietf.org
> >> To: IETF-Announce: ;
> >>
> >> A New Internet-Draft is available from the on-line Internet-Drafts
> > directories.
> >>
> >>
> >> Title : Real-Time Facsimile (T.38) - audio/t38 MIME Sub-type
> >>                            Registration
> >> Author(s) : P. Jones
> >> Filename : draft-jones-avt-audio-t38-00.txt
> >> Pages : 4
> >> Date : 2003-8-11
> >>
> >> This document defines the MIME sub-type audio/t38.  The packetization
> >> and usage of this MIME type, which is intended for use within SDP [2]
> >> and H.248 [3] systems, is specified within ITU-T Recommendation T.38
> >> [1].
> >>
> >> A URL for this Internet-Draft is:
> >> http://www.ietf.org/internet-drafts/draft-jones-avt-audio-t38-00.txt
> >>
> >> To remove yourself from the IETF Announcement list, send a message to
> >> ietf-announce-request with the word unsubscribe in the body of the
> > message.
> >>
> >> Internet-Drafts are also available by anonymous FTP. Login with the
> > username
> >> "anonymous" and a password of your e-mail address. After logging in,
> >> type "cd internet-drafts" and then
> >> "get draft-jones-avt-audio-t38-00.txt".
> >>
> >> A list of Internet-Drafts directories can be found in
> >> http://www.ietf.org/shadow.html
> >> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >> ...
> >>
> >
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www1.ietf.org/mailman/listinfo/avt
> >
> >
> -- 
> Colin Perkins
> http://csperkins.org/
>


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] Progressing ULP and UXP
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Folks,

The working group has previously agreed that the drafts on Generic FEC 
with Uneven Level Protection (draft-ietf-avt-ulp-07.txt) and 
Erasure-Resilient Transmission of Progressive Multimedia Streams 
(draft-ietf-avt-uxp-05.txt) provide somewhat similar services, but are 
targeted at different environments or requirements. We proposed that 
the drafts should advance to the IESG together, with applicability 
statements showing the differing emphasis of the proposals.

Since that time, the ULP draft has broadened in scope to become a 
proposed replacement for the Parity FEC specification of RFC 2733. The 
UXP draft remains narrowly targeted, and has been blocked waiting for 
the completion of the ULP work.

Accordingly, due to the increased scope of the ULP draft, we now 
propose to allow these drafts to advance separately to completion, 
rather than linking their progress (neither draft is complete at this 
time).

We welcome any comments from the working group on this proposal.

Thanks,
Colin and Magnus


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Deepak Kotian <dkotian3@vsnl.net>
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Hi,

Can I please get a good reference documents for RTP timestamp calculation for MPEG1 Video and Audio ES so that it works with RTSP client like RealOne/QuickTime,
Basically, based on RFC 1889 and RFC 2250.

Any help on this would be appreciated.

Thanks and Regard
Deepak




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<DIV><FONT face=Arial size=2>Hi,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>Can I please get a good reference documents for 
</FONT><FONT face=Arial size=2><FONT size=1>RTP timestamp calculation for MPEG1 
Video and Audio ES so that it works with RTSP client like 
RealOne/QuickTime,</FONT></FONT></DIV>
<DIV><FONT face=Arial size=2><FONT size=1>Basically, based on RFC 1889 and RFC 
2250.</FONT></FONT></DIV>
<DIV><FONT face=Arial size=2><FONT size=1></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial size=2><FONT size=1><FONT size=2>Any help on this would be 
appreciated.</FONT></FONT></FONT></DIV>
<DIV><FONT face=Arial size=2><FONT size=1><FONT 
size=2></FONT></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial size=2><FONT size=1><FONT size=2>Thanks and 
Regard</FONT></FONT></FONT></DIV>
<DIV><FONT face=Arial size=2><FONT size=1><FONT 
size=2>Deepak</FONT></FONT></FONT></DIV>
<DIV><FONT face=Arial size=2><FONT size=1>&nbsp;</DIV>
<P>&nbsp;</P></FONT></FONT></BODY></HTML>

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_______________________________________________
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Subject: Re: [AVT] RTP timestamp calculation for MPEG1 Video and Audio ES so that itworks with RTSP client like RealOne/QuickTime
CC: avt@ietf.org
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<html><head><meta Name='keywords' Content='commtouch, pronto, mail, free email, free, branded, web based, free web based email, communications, internet, software, advertising banners, e-mail, free software'></head><body   ><div align='left'><font   ><blockquote><blockquote><TT>Hi,<BR>
<BR>
 Since you have mentioned the documents yourself, I think they <BR>
provide sufficient information for interop with realone. Check out <BR>
rfc3550 too.<BR>
<BR>
 We need to provide the timestamp with the proper clock rates <BR>
mentioned for the specific media type payload format (rfc2250).<BR>
<BR>
 For example, if the clock is 90kHz, practically, multiply the <BR>
timestamp in seconds by 90000. This is the RTP &nbsp;timestamp value to be <BR>
transmitted in the packet. Check out the clock rates carefully for <BR>
each media type and associated payload format is used.<BR>
<BR>
 Take care of putting proper NTP and RTP timestamp correlation while <BR>
sending SR packets.<BR>
<BR>
 There is also an interoperablity statement for RTP/RTSP in the real <BR>
networks site. Sorry, I am not able to cite the exact document at <BR>
present. try downloading from their website.<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
</TT><br><br><font><p align=left><TT>--Neo</TT><br>Get your own Free E-mail at Witch Wisdom http://www.witchwisdom.com<br>__________________<br>http://www.zzn.com</blockquote></blockquote></div></font></body></html>

_______________________________________________
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From: Deepak Kotian <dkotian3@vsnl.net>
To: avt@ietf.org
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Subject: [AVT] Where can I get ISO/IEC 1172 from
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Hi,

    I want to implement RFC 2250 for MPEG1 Video and Audio ES.
    Helix( RealOne forum) say that I would need ISO/IEC 1172 for RFC=20
    implementation. Where can I get ISO/IEC 1172 from, any inputs on =
this one please

Here is the extract of the document from real networks
***********************
=B7       MPEG-1
Helix Universal Server delivers ISO/IEC 1172 compliant video and system =
bit=20
streams over RTSP/RTP. RealOne Player and QuickTime Player, for example, =

can play MPEG-1 streams. Helix Universal Server delivers MPEG-1 content=20
with file extensions of mpa, mpg, mpeg, mpv, mps, m2v, m1v, and mpe
***********************

Thanks and Regards
Deepak

  ----- Original Message -----=20
  From: Deepak Kotian=20
  To: Neo@wiseone.zzn.com=20
  Cc: avt@ietf.org=20
  Sent: Tuesday, August 26, 2003 8:26 PM
  Subject: Re: [AVT] RTP timestamp calculation for MPEG1 Video and Audio =
ES so that itworks with RTSP client like RealOne/QuickTime


  Hi,

  >>There is also an interoperablity statement for >>RTP/RTSP in the =
real=20
  >>networks site. Sorry, I am not able to cite the >>exact document at=20
  >>present. try downloading from their website.

  Just to add.
  About this documents, there seems to be no=20
  document with the RealOne guys for the timestamp calculation,etc. I =
checked with them at the helix forum.

  Thanks and Regards
  Deepak


  ----- Original Message -----=20
    From: Deepak Kotian=20
    To: Neo@wiseone.zzn.com=20
    Cc: avt@ietf.org=20
    Sent: Tuesday, August 26, 2003 7:33 PM
    Subject: Re: [AVT] RTP timestamp calculation for MPEG1 Video and =
Audio ES so that itworks with RTSP client like RealOne/QuickTime


    Hi,

    Thanks for the inputs.
    But, these RFCs do not mention about the PTS or DTS timestamp
    calculation, it seems the RTP timestamp is derived from PTS or DTS
    of headers. Which is usually 33 bits, that too is mystery to me as
    the RTP timestamp is 32 bits.=20
    That is what I have heard.

    Is there a very simple implementation where Minimal RTSP server
    is implemented and works with RealOne via RTP for both MPEG1 audio=20
    and video Elementary stream.
    The helix source is really very big to understand quickly.
    Some more inputs would be of great help.

    Thanks and Regards
    Deepak


      ----- Original Message -----=20
      From: Neo Anderson=20
      To: dkotian3@vsnl.net=20
      Cc: avt@ietf.org=20
      Sent: Tuesday, August 26, 2003 11:14 AM
      Subject: Re: [AVT] RTP timestamp calculation for MPEG1 Video and =
Audio ES so that itworks with RTSP client like RealOne/QuickTime


          Hi,

          Since you have mentioned the documents yourself, I think they=20
          provide sufficient information for interop with realone. Check =
out=20
          rfc3550 too.

          We need to provide the timestamp with the proper clock rates=20
          mentioned for the specific media type payload format =
(rfc2250).

          For example, if the clock is 90kHz, practically, multiply the=20
          timestamp in seconds by 90000. This is the RTP  timestamp =
value to be=20
          transmitted in the packet. Check out the clock rates carefully =
for=20
          each media type and associated payload format is used.

          Take care of putting proper NTP and RTP timestamp correlation =
while=20
          sending SR packets.

          There is also an interoperablity statement for RTP/RTSP in the =
real=20
          networks site. Sorry, I am not able to cite the exact document =
at=20
          present. try downloading from their website.









          --Neo
          Get your own Free E-mail at Witch Wisdom =
http://www.witchwisdom.com
          __________________
          http://www.zzn.com


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META=20
content=3D"commtouch, pronto, mail, free email, free, branded, web =
based, free web based email, communications, internet, software, =
advertising banners, e-mail, free software"=20
name=3Dkeywords>
<META content=3D"MSHTML 5.50.4522.1801" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; I want to implement =
RFC 2250 for=20
MPEG1 Video and Audio ES.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; Helix(&nbsp;RealOne=20
forum)&nbsp;say that I would need <FONT face=3D"Times New Roman" =
size=3D3>ISO/IEC=20
1172 for RFC </FONT></FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; implementation. =
Where can I get=20
<FONT face=3D"Times New Roman" size=3D3>ISO/IEC 1172 </FONT>from, any =
inputs on this=20
one please</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Here is the extract of the document =
from real=20
networks</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>***********************</FONT></DIV>
<DIV>=B7&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; MPEG-1<BR>Helix Universal =
Server=20
delivers ISO/IEC 1172 compliant video and system bit <BR>streams over =
RTSP/RTP.=20
RealOne Player and QuickTime Player, for example, <BR>can play MPEG-1 =
streams.=20
Helix Universal Server delivers MPEG-1 content <BR>with file extensions =
of mpa,=20
mpg, mpeg, mpv, mps, m2v, m1v, and mpe</DIV>
<DIV><FONT face=3DArial size=3D2>***********************</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks and Regards</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Deepak</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
  <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV=20
  style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
  <A title=3Ddkotian3@vsnl.net href=3D"mailto:dkotian3@vsnl.net">Deepak =
Kotian</A>=20
  </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A =
title=3DNeo@wiseone.zzn.com=20
  href=3D"mailto:Neo@wiseone.zzn.com">Neo@wiseone.zzn.com</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Cc:</B> <A title=3Davt@ietf.org=20
  href=3D"mailto:avt@ietf.org">avt@ietf.org</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Tuesday, August 26, 2003 =
8:26=20
  PM</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> Re: [AVT] RTP =
timestamp=20
  calculation for MPEG1 Video and Audio ES so that itworks with RTSP =
client like=20
  RealOne/QuickTime</DIV>
  <DIV><BR></DIV>
  <DIV><FONT face=3DArial size=3D2>Hi,</FONT></DIV>
  <DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><FONT face=3D"Courier New" =
size=3D3>&gt;&gt;There is=20
  also an interoperablity statement for &gt;&gt;RTP/RTSP in the real=20
  <BR>&gt;&gt;networks site. Sorry, I am not able to cite the =
&gt;&gt;exact=20
  document at <BR>&gt;&gt;present. try downloading from their=20
  website.</FONT></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><FONT face=3D"Courier New"=20
  size=3D3></FONT></FONT>&nbsp;</DIV>
  <DIV><FONT face=3D"Courier New">Just to add.</FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><FONT face=3D"Courier New" =
size=3D3>About this=20
  documents, there seems to be no </FONT></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><FONT face=3D"Courier New" =
size=3D3>document with the=20
  RealOne guys for the timestamp calculation,etc. I checked with them at =
the=20
  helix forum.</FONT></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><FONT face=3D"Courier New"=20
  size=3D3></FONT></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><FONT face=3D"Courier New" =
size=3D3>Thanks and=20
  Regards</FONT></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><FONT face=3D"Courier New"=20
  size=3D3>Deepak</FONT></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2>&nbsp;</DIV>
  <DIV><BR></FONT>----- Original Message ----- </DIV>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
    <DIV=20
    style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
    <A title=3Ddkotian3@vsnl.net =
href=3D"mailto:dkotian3@vsnl.net">Deepak Kotian</A>=20
    </DIV>
    <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A =
title=3DNeo@wiseone.zzn.com=20
    href=3D"mailto:Neo@wiseone.zzn.com">Neo@wiseone.zzn.com</A> </DIV>
    <DIV style=3D"FONT: 10pt arial"><B>Cc:</B> <A title=3Davt@ietf.org=20
    href=3D"mailto:avt@ietf.org">avt@ietf.org</A> </DIV>
    <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Tuesday, August 26, =
2003 7:33=20
    PM</DIV>
    <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> Re: [AVT] RTP =
timestamp=20
    calculation for MPEG1 Video and Audio ES so that itworks with RTSP =
client=20
    like RealOne/QuickTime</DIV>
    <DIV><BR></DIV>
    <DIV><FONT face=3DArial size=3D2>Hi,</FONT></DIV>
    <DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial size=3D2>Thanks for the inputs.</FONT></DIV>
    <DIV><FONT face=3DArial size=3D2>But, these RFCs do not mention =
about the PTS or=20
    DTS timestamp</FONT></DIV>
    <DIV><FONT face=3DArial size=3D2>calculation, it seems the RTP =
timestamp is=20
    derived from PTS or DTS</FONT></DIV>
    <DIV><FONT face=3DArial size=3D2>of headers. Which is usually 33 =
bits, that too=20
    is mystery to me as</FONT></DIV>
    <DIV><FONT face=3DArial size=3D2>the RTP timestamp is 32=20
bits.</FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial size=3D2>That is what I have =
heard.</FONT></DIV>
    <DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial size=3D2>Is there a very simple =
implementation where=20
    Minimal RTSP server</FONT></DIV>
    <DIV><FONT face=3DArial size=3D2>is implemented and works with =
RealOne via RTP=20
    for both MPEG1 audio </FONT></DIV>
    <DIV><FONT face=3DArial size=3D2>and video Elementary =
stream.</FONT></DIV>
    <DIV><FONT face=3DArial size=3D2>The helix source is really very big =
to=20
    understand quickly.</FONT></DIV>
    <DIV><FONT face=3DArial size=3D2>Some more inputs would be of great=20
    help.</FONT></DIV>
    <DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial size=3D2>Thanks and Regards</FONT></DIV>
    <DIV><FONT face=3DArial size=3D2>Deepak</FONT></DIV>
    <DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
    <BLOCKQUOTE dir=3Dltr=20
    style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
      <DIV style=3D"FONT: 10pt arial">----- Original Message ----- =
</DIV>
      <DIV=20
      style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
      <A title=3Dneo@wiseone.zzn.com =
href=3D"mailto:neo@wiseone.zzn.com">Neo=20
      Anderson</A> </DIV>
      <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A =
title=3Ddkotian3@vsnl.net=20
      href=3D"mailto:dkotian3@vsnl.net">dkotian3@vsnl.net</A> </DIV>
      <DIV style=3D"FONT: 10pt arial"><B>Cc:</B> <A title=3Davt@ietf.org =

      href=3D"mailto:avt@ietf.org">avt@ietf.org</A> </DIV>
      <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Tuesday, August 26, =
2003 11:14=20
      AM</DIV>
      <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> Re: [AVT] RTP =
timestamp=20
      calculation for MPEG1 Video and Audio ES so that itworks with RTSP =
client=20
      like RealOne/QuickTime</DIV>
      <DIV><BR></DIV>
      <DIV align=3Dleft><FONT size=3D+0>
      <BLOCKQUOTE>
        <BLOCKQUOTE><TT>Hi,<BR><BR>Since you have mentioned the =
documents=20
          yourself, I think they <BR>provide sufficient information for =
interop=20
          with realone. Check out <BR>rfc3550 too.<BR><BR>We need to =
provide the=20
          timestamp with the proper clock rates <BR>mentioned for the =
specific=20
          media type payload format (rfc2250).<BR><BR>For example, if =
the clock=20
          is 90kHz, practically, multiply the <BR>timestamp in seconds =
by 90000.=20
          This is the RTP &nbsp;timestamp value to be <BR>transmitted in =
the=20
          packet. Check out the clock rates carefully for <BR>each media =
type=20
          and associated payload format is used.<BR><BR>Take care of =
putting=20
          proper NTP and RTP timestamp correlation while <BR>sending SR=20
          packets.<BR><BR>There is also an interoperablity statement for =

          RTP/RTSP in the real <BR>networks site. Sorry, I am not able =
to cite=20
          the exact document at <BR>present. try downloading from their=20
          website.<BR><BR><BR><BR><BR><BR><BR></TT><BR><BR><FONT =
size=3D+0>
          <P align=3Dleft><TT>--Neo</TT><BR>Get your own Free E-mail at =
Witch=20
          Wisdom=20
          =
http://www.witchwisdom.com<BR>__________________<BR>http://www.zzn.com</P=
></BLOCKQUOTE></BLOCKQUOTE></DIV></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE><=
/FONT></FONT></BODY></HTML>

--Boundary_(ID_B/F6T02gIujN0dfkvFJ71Q)--

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Tue, 26 Aug 2003 13:27:14 -0700 (PDT)
From: Stephen Casner <casner@acm.org>
To: Henning Schulzrinne <hgs@cs.columbia.edu>
cc: Oren Peleg <OrenP@audiocodes.com>, avt@ietf.org
In-Reply-To: <3F4B5BD6.9040908@cs.columbia.edu>
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 <3F4B5BD6.9040908@cs.columbia.edu>
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Subject: [AVT] Re: Question on 4.1 section of RFC3551
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On Tue, 26 Aug 2003, Henning Schulzrinne wrote:

> For example, in a conference bridge, it would be counterproductive to
> send CN packets for all but the active speaker(s). Also, IETF specs
> aren't product sales and training manuals - it is not the IETF's job to
> tell people that engineering systems without CN is generally a bad idea.
> Sending CN is not required for interoperability, so being too
> prescriptive doesn't seem helpful here.

I agree with Henning's sentiments here.  To go further, I would say
that the receiver SHOULD be prepared to work well even if CN is not
received, because it is always possible that the CN packet may be lost
even if it was sent.

Many miles were logged on the MBone tools (vat, RAT, NeVoT and
friends) with no CN.  You may choose to discount these as not be of
service grade, but they worked properly without requiring CN.  When we
have multicast sessions from the IETF meeting with hundreds of
participants, we were glad that those hundreds of participants weren't
sending CN all the time.  Yet at any point that any one of them wanted
to say something back to the session, the communication path was
ready.

                                                        -- Steve

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Tue, 26 Aug 2003 20:26:19 +0530
From: Deepak Kotian <dkotian3@vsnl.net>
Subject: Re: [AVT] RTP timestamp calculation for MPEG1 Video and Audio ES so
 that itworks with RTSP client like RealOne/QuickTime
To: Neo@wiseone.zzn.com
Cc: avt@ietf.org
Reply-to: Deepak Kotian <dkotian3@vsnl.net>
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Hi,

>>There is also an interoperablity statement for >>RTP/RTSP in the real 
>>networks site. Sorry, I am not able to cite the >>exact document at 
>>present. try downloading from their website.

Just to add.
About this documents, there seems to be no 
document with the RealOne guys for the timestamp calculation,etc. I checked with them at the helix forum.

Thanks and Regards
Deepak


----- Original Message ----- 
  From: Deepak Kotian 
  To: Neo@wiseone.zzn.com 
  Cc: avt@ietf.org 
  Sent: Tuesday, August 26, 2003 7:33 PM
  Subject: Re: [AVT] RTP timestamp calculation for MPEG1 Video and Audio ES so that itworks with RTSP client like RealOne/QuickTime


  Hi,

  Thanks for the inputs.
  But, these RFCs do not mention about the PTS or DTS timestamp
  calculation, it seems the RTP timestamp is derived from PTS or DTS
  of headers. Which is usually 33 bits, that too is mystery to me as
  the RTP timestamp is 32 bits. 
  That is what I have heard.

  Is there a very simple implementation where Minimal RTSP server
  is implemented and works with RealOne via RTP for both MPEG1 audio 
  and video Elementary stream.
  The helix source is really very big to understand quickly.
  Some more inputs would be of great help.

  Thanks and Regards
  Deepak


    ----- Original Message ----- 
    From: Neo Anderson 
    To: dkotian3@vsnl.net 
    Cc: avt@ietf.org 
    Sent: Tuesday, August 26, 2003 11:14 AM
    Subject: Re: [AVT] RTP timestamp calculation for MPEG1 Video and Audio ES so that itworks with RTSP client like RealOne/QuickTime


        Hi,

        Since you have mentioned the documents yourself, I think they 
        provide sufficient information for interop with realone. Check out 
        rfc3550 too.

        We need to provide the timestamp with the proper clock rates 
        mentioned for the specific media type payload format (rfc2250).

        For example, if the clock is 90kHz, practically, multiply the 
        timestamp in seconds by 90000. This is the RTP  timestamp value to be 
        transmitted in the packet. Check out the clock rates carefully for 
        each media type and associated payload format is used.

        Take care of putting proper NTP and RTP timestamp correlation while 
        sending SR packets.

        There is also an interoperablity statement for RTP/RTSP in the real 
        networks site. Sorry, I am not able to cite the exact document at 
        present. try downloading from their website.









        --Neo
        Get your own Free E-mail at Witch Wisdom http://www.witchwisdom.com
        __________________
        http://www.zzn.com


--Boundary_(ID_RupAAvP1cosB7xO7InEKdA)
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<META 
content="commtouch, pronto, mail, free email, free, branded, web based, free web based email, communications, internet, software, advertising banners, e-mail, free software" 
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<DIV><FONT face=Arial size=2>Hi,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial size=2><FONT face="Courier New" size=3>&gt;&gt;There is 
also an interoperablity statement for &gt;&gt;RTP/RTSP in the real 
<BR>&gt;&gt;networks site. Sorry, I am not able to cite the &gt;&gt;exact 
document at <BR>&gt;&gt;present. try downloading from their 
website.</FONT></FONT></DIV>
<DIV><FONT face=Arial size=2><FONT face="Courier New" 
size=3></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face="Courier New">Just to add.</FONT></DIV>
<DIV><FONT face=Arial size=2><FONT face="Courier New" size=3>About this 
documents, there seems to be no </FONT></FONT></DIV>
<DIV><FONT face=Arial size=2><FONT face="Courier New" size=3>document with the 
RealOne guys for the timestamp calculation,etc. I checked with them at the helix 
forum.</FONT></FONT></DIV>
<DIV><FONT face=Arial size=2><FONT face="Courier New" 
size=3></FONT></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial size=2><FONT face="Courier New" size=3>Thanks and 
Regards</FONT></FONT></DIV>
<DIV><FONT face=Arial size=2><FONT face="Courier New" 
size=3>Deepak</FONT></FONT></DIV>
<DIV><FONT face=Arial size=2>&nbsp;</DIV>
<DIV><BR></FONT>----- Original Message ----- </DIV>
<BLOCKQUOTE dir=ltr 
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
  <DIV 
  style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B> 
  <A title=dkotian3@vsnl.net href="mailto:dkotian3@vsnl.net">Deepak Kotian</A> 
  </DIV>
  <DIV style="FONT: 10pt arial"><B>To:</B> <A title=Neo@wiseone.zzn.com 
  href="mailto:Neo@wiseone.zzn.com">Neo@wiseone.zzn.com</A> </DIV>
  <DIV style="FONT: 10pt arial"><B>Cc:</B> <A title=avt@ietf.org 
  href="mailto:avt@ietf.org">avt@ietf.org</A> </DIV>
  <DIV style="FONT: 10pt arial"><B>Sent:</B> Tuesday, August 26, 2003 7:33 
  PM</DIV>
  <DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [AVT] RTP timestamp 
  calculation for MPEG1 Video and Audio ES so that itworks with RTSP client like 
  RealOne/QuickTime</DIV>
  <DIV><BR></DIV>
  <DIV><FONT face=Arial size=2>Hi,</FONT></DIV>
  <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial size=2>Thanks for the inputs.</FONT></DIV>
  <DIV><FONT face=Arial size=2>But, these RFCs do not mention about the PTS or 
  DTS timestamp</FONT></DIV>
  <DIV><FONT face=Arial size=2>calculation, it seems the RTP timestamp is 
  derived from PTS or DTS</FONT></DIV>
  <DIV><FONT face=Arial size=2>of headers. Which is usually 33 bits, that too is 
  mystery to me as</FONT></DIV>
  <DIV><FONT face=Arial size=2>the RTP timestamp is 32 bits.</FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial size=2>That is what I have heard.</FONT></DIV>
  <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial size=2>Is there a very simple implementation where 
  Minimal RTSP server</FONT></DIV>
  <DIV><FONT face=Arial size=2>is implemented and works with RealOne via RTP for 
  both MPEG1 audio </FONT></DIV>
  <DIV><FONT face=Arial size=2>and video Elementary stream.</FONT></DIV>
  <DIV><FONT face=Arial size=2>The helix source is really very big to understand 
  quickly.</FONT></DIV>
  <DIV><FONT face=Arial size=2>Some more inputs would be of great 
  help.</FONT></DIV>
  <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial size=2>Thanks and Regards</FONT></DIV>
  <DIV><FONT face=Arial size=2>Deepak</FONT></DIV>
  <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
  <BLOCKQUOTE dir=ltr 
  style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
    <DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
    <DIV 
    style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B> 
    <A title=neo@wiseone.zzn.com href="mailto:neo@wiseone.zzn.com">Neo 
    Anderson</A> </DIV>
    <DIV style="FONT: 10pt arial"><B>To:</B> <A title=dkotian3@vsnl.net 
    href="mailto:dkotian3@vsnl.net">dkotian3@vsnl.net</A> </DIV>
    <DIV style="FONT: 10pt arial"><B>Cc:</B> <A title=avt@ietf.org 
    href="mailto:avt@ietf.org">avt@ietf.org</A> </DIV>
    <DIV style="FONT: 10pt arial"><B>Sent:</B> Tuesday, August 26, 2003 11:14 
    AM</DIV>
    <DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [AVT] RTP timestamp 
    calculation for MPEG1 Video and Audio ES so that itworks with RTSP client 
    like RealOne/QuickTime</DIV>
    <DIV><BR></DIV>
    <DIV align=left><FONT size=+0>
    <BLOCKQUOTE>
      <BLOCKQUOTE><TT>Hi,<BR><BR>Since you have mentioned the documents 
        yourself, I think they <BR>provide sufficient information for interop 
        with realone. Check out <BR>rfc3550 too.<BR><BR>We need to provide the 
        timestamp with the proper clock rates <BR>mentioned for the specific 
        media type payload format (rfc2250).<BR><BR>For example, if the clock is 
        90kHz, practically, multiply the <BR>timestamp in seconds by 90000. This 
        is the RTP &nbsp;timestamp value to be <BR>transmitted in the packet. 
        Check out the clock rates carefully for <BR>each media type and 
        associated payload format is used.<BR><BR>Take care of putting proper 
        NTP and RTP timestamp correlation while <BR>sending SR 
        packets.<BR><BR>There is also an interoperablity statement for RTP/RTSP 
        in the real <BR>networks site. Sorry, I am not able to cite the exact 
        document at <BR>present. try downloading from their 
        website.<BR><BR><BR><BR><BR><BR><BR></TT><BR><BR><FONT size=+0>
        <P align=left><TT>--Neo</TT><BR>Get your own Free E-mail at Witch Wisdom 
        http://www.witchwisdom.com<BR>__________________<BR>http://www.zzn.com</P></BLOCKQUOTE></BLOCKQUOTE></DIV></BLOCKQUOTE></BLOCKQUOTE></FONT></FONT></BODY></HTML>

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_______________________________________________
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Thread-Topic: Question on 4.1 section of RFC3551
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From: "Oren Peleg" <OrenP@audiocodes.com>
To: <schulzrinne@cs.columbia.edu>, <casner@acm.org>
Cc: <avt@ietf.org>
Subject: [AVT] Question on 4.1 section of RFC3551
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Hi,

At 4.1 section of the RFC3551 it is written:
"Some payload formats (see Sections 4.5.3 and 4.5.6) define a "silence =
insertion descriptor" or "comfort noise" frame to specify parameters for =
artificial noise that may be generated during a period of silence to =
approximate the background noise at the source. For other payload =
formats, a generic Comfort Noise (CN) payload format is specified in RFC =
3389 [9]. When the CN payload format is used with another payload =
format, different values in the RTP payload type field distinguish =
comfort-noise packets from those of the selected payload format."
Shouldnt it be mentioned that a CN packet (of any format) SHOULD be used =
on silence detection?
There is a reference to "applications which send either no packets or =
occasional comfort- noise packets during silence" Since G711 AppendixII =
specify CN for G711 (and even recommend using in other coders without =
Built-in Silence compression), What kind of application is it that does =
not send any packet during Silence? isnt it contradicts the above =
section which suggests that the RFC3389 is a valid option for payload =
format who does not have CN capabilities?
Thanks,

Oren Peleg

Work phone No. : 03 - 5394167
Home phone No. : 03 - 5335913
Cell phone No. : 055 - 669613
Email : Oren.Peleg@audiocodes.com


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<TITLE>Question on 4.1 section of RFC3551</TITLE>
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<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">Hi,</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">At 4.1 section of the RFC3551 =
it is written:</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">&quot;</FONT><FONT =
FACE=3D"Times New Roman">Some payload formats (see Sections 4.5.3 and =
4.5.6) define a &quot;silence insertion descriptor&quot; or =
&quot;comfort noise&quot; frame to specify parameters for artificial =
noise that may be generated during a period of silence to approximate =
the background noise at the source. For other payload formats, a generic =
Comfort Noise (CN) payload format is specified in RFC 3389 [9]. When the =
CN payload format is used with another payload format, different values =
in the RTP payload type field distinguish comfort-noise packets from =
those of the selected payload format.&quot;</FONT></P>

<P DIR=3DLTR><FONT FACE=3D"Times New Roman">Shouldnt it be mentioned =
that a CN packet (of any format) SHOULD be used on silence =
detection?</FONT></P>

<P DIR=3DLTR><FONT FACE=3D"Times New Roman">There is a reference to =
&quot;applications which send either no packets or occasional comfort- =
noise packets during silence&quot; Since G711 AppendixII specify CN for =
G711 (and even recommend using in other coders without Built-in Silence =
compression), What kind of application is it that does not send any =
packet during Silence? isnt it contradicts the above section which =
suggests that the RFC3389 is a valid option for payload format who does =
not have CN capabilities?</FONT></P>

<P DIR=3DLTR><FONT FACE=3D"Times New Roman">Thanks,</FONT></P>

<P DIR=3DLTR><FONT COLOR=3D"#000080" SIZE=3D4 FACE=3D"Monotype =
Corsiva">Oren Peleg</FONT></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Book Antiqua">Work phone No. : 03 - 5394167</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Book Antiqua">Home phone No. : 03 - 5335913</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Book Antiqua">Cell phone No. : 055 - 669613</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Book =
Antiqua">Email :<U> </U></FONT><U><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Book =
Antiqua">Oren.Peleg@audiocodes.com</FONT></U></SPAN><U><SPAN =
LANG=3D"he"></SPAN></U><SPAN LANG=3D"he"></SPAN></P>

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_______________________________________________
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https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Aug 27 02:45:00 2003
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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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Subject: [AVT] Re: Question on 4.1 section of RFC3551
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For example, in a conference bridge, it would be counterproductive to 
send CN packets for all but the active speaker(s). Also, IETF specs 
aren't product sales and training manuals - it is not the IETF's job to 
tell people that engineering systems without CN is generally a bad idea. 
Sending CN is not required for interoperability, so being too 
prescriptive doesn't seem helpful here.

Oren Peleg wrote:

> Hi,
> 
> At 4.1 section of the RFC3551 it is written:
> 
> "Some payload formats (see Sections 4.5.3 and 4.5.6) define a "silence 
> insertion descriptor" or "comfort noise" frame to specify parameters for 
> artificial noise that may be generated during a period of silence to 
> approximate the background noise at the source. For other payload 
> formats, a generic Comfort Noise (CN) payload format is specified in RFC 
> 3389 [9]. When the CN payload format is used with another payload 
> format, different values in the RTP payload type field distinguish 
> comfort-noise packets from those of the selected payload format."
> 
> Shouldnt it be mentioned that a CN packet (of any format) SHOULD be used 
> on silence detection?
> 
> There is a reference to "applications which send either no packets or 
> occasional comfort- noise packets during silence" Since G711 AppendixII 
> specify CN for G711 (and even recommend using in other coders without 
> Built-in Silence compression), What kind of application is it that does 
> not send any packet during Silence? isnt it contradicts the above 
> section which suggests that the RFC3389 is a valid option for payload 
> format who does not have CN capabilities?
> 
> Thanks,
> 
> Oren Peleg
> 
> Work phone No. : 03 - 5394167
> 
> Home phone No. : 03 - 5335913
> 
> Cell phone No. : 055 - 669613
> 
> Email : Oren.Peleg@audiocodes.com
> 


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Aug 27 13:49:42 2003
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Subject: [AVT] Interpreting RFC2327: SDP whitespace problem in "s=" line
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Hello All:

During recent interop testing, one issue has come up regarding whether it is
valid to have "s=[single space]" in
an SDP as the subject of a presentation.

In section 6 of RFC2327 (see
http://www.ietf.org/rfc/rfc2327.txt?number=2327):

   An SDP session description consists of a number of lines of text of
   the form <type>=<value> <type> is always exactly one character and is
   case-significant.  <value> is a structured text string whose format
   depends on <type>.  It also will be case-significant unless a
   specific field defines otherwise.  Whitespace is not permitted either
   side of the `=' sign. In general <value> is either a number of fields
   delimited by a single space character or a free format string.

Taken off context, this paragraph seems to be prohibiting
'<type>[space]=<value>'  or
'<type>=[space]<value>' .  However, we don't think the intent was to place
restrictions on <value>.  If <value> is a free-format string, then it should
be able to start with a space. (And this is what the ABNF definition says,
for the value in the 's=' case).

We checked
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-new-13.txt as
well, which clearly states that
if there is no meaningful subject, a single space should be used in "s="
line, yet it apparently has contracdictory statement in section 5.

We hereby would like to confirm with this mailing list, that "s=[single
space]" is a legal subject line. 
We would also appreciate clarification in
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-new-13.txt , 
section 5, to indicate that:

   " An SDP session description consists of a number of lines of text of
   the form:

      <type>=<value>

   where <type> MUST be exactly one case-significant character and
   <value> is structured text whose format depends on <type>.  In
   general <value> is either a number of fields delimited by a single
   space character, or a free format string.   [modified text]  Whitespace
MUST NOT be used either side of the "=" sign, except if the free    format
<vallue> string  begins with a space or contains a single space. An exampel
is: "s=[single space]", which is a   legal SDP line."

Thanks in advance for your feedback.

thomas
PacketVideo Corp.

www.pv.com





_______________________________________________
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This is a multi-part message in MIME format.

--Boundary_(ID_oxjF1x8snIqivsFUXbmBYw)
Content-type: text/plain; charset=iso-8859-1
Content-Transfer-Encoding: quoted-printable

Thanks for the input.
So, basically it is difficult to get it.
Can someone in the forum have a handy document for calculation of RTP =
timestamp with reference
to PTS/DTS for Video and Audio ES.

Any help on this would be great.

Thanks and Regards
Deepak
  ----- Original Message -----=20
  From: Sandy (Alexander) MacInnis=20
  To: Deepak Kotian ; avt@ietf.org=20
  Sent: Wednesday, August 27, 2003 6:09 AM
  Subject: Re: [AVT] Where can I get ISO/IEC 1172 from


  The MPEG-1 standard is ISO/IEC 11172 (not 1172). That looks like a =
typo.

  The systems part is 11172-1, video is 11172-2, and audio is 11172-3.

  These standards are copyrighted by ISO/IEC, and they are available in =
hard copy from the national standards bodies and IEC.

  There may be some people with bootleg non-final soft copies, but as =
far as I know they are not permitted to be distributed due to copyright.

  --Sandy MacInnis
  Broadcom
  (Chair of MPEG-1 Systems and MPEG-2 Systems at the time they were =
developed.)

  At 02:42 PM 8/26/2003, Deepak Kotian wrote:

    Hi,
    =20
        I want to implement RFC 2250 for MPEG1 Video and Audio ES.
        Helix( RealOne forum) say that I would need ISO/IEC 1172 for RFC =

        implementation. Where can I get ISO/IEC 1172 from, any inputs on =
this one please
    =20
    Here is the extract of the document from real networks
    ***********************
    =B7       MPEG-1
    Helix Universal Server delivers ISO/IEC 1172 compliant video and =
system bit=20
    streams over RTSP/RTP. RealOne Player and QuickTime Player, for =
example,=20
    can play MPEG-1 streams. Helix Universal Server delivers MPEG-1 =
content=20
    with file extensions of mpa, mpg, mpeg, mpv, mps, m2v, m1v, and mpe
    ***********************
    =20
    Thanks and Regards
    Deepak

--Boundary_(ID_oxjF1x8snIqivsFUXbmBYw)
Content-type: text/html; charset=iso-8859-1
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 5.50.4522.1801" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Thanks for the input.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>So, basically it is difficult to get=20
it.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Can someone in the forum have a handy =
document for=20
calculation of RTP timestamp with reference</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>to PTS/DTS for Video and Audio =
ES.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Any help on this would be =
great.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks and Regards</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Deepak</FONT></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
  <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV=20
  style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
  <A title=3Dmacinnis@broadcom.com =
href=3D"mailto:macinnis@broadcom.com">Sandy=20
  (Alexander) MacInnis</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A =
title=3Ddkotian3@vsnl.net=20
  href=3D"mailto:dkotian3@vsnl.net">Deepak Kotian</A> ; <A =
title=3Davt@ietf.org=20
  href=3D"mailto:avt@ietf.org">avt@ietf.org</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Wednesday, August 27, =
2003 6:09=20
  AM</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> Re: [AVT] Where can I =
get=20
  ISO/IEC 1172 from</DIV>
  <DIV><BR></DIV>The MPEG-1 standard is ISO/IEC 11172 (not 1172). That =
looks=20
  like a typo.<BR><BR>The systems part is 11172-1, video is 11172-2, and =
audio=20
  is 11172-3.<BR><BR>These standards are copyrighted by ISO/IEC, and =
they are=20
  available in hard copy from the national standards bodies and=20
  IEC.<BR><BR>There may be some people with bootleg non-final soft =
copies, but=20
  as far as I know they are not permitted to be distributed due to=20
  copyright.<BR><BR>--Sandy MacInnis<BR>Broadcom<BR>(Chair of MPEG-1 =
Systems and=20
  MPEG-2 Systems at the time they were developed.)<BR><BR>At 02:42 PM =
8/26/2003,=20
  Deepak Kotian wrote:<BR>
  <BLOCKQUOTE cite type=3D"cite"><FONT face=3Darial=20
    size=3D2>Hi,</FONT><BR>&nbsp;<BR><FONT face=3Darial =
size=3D2>&nbsp;&nbsp;&nbsp; I=20
    want to implement RFC 2250 for MPEG1 Video and Audio =
ES.</FONT><BR><FONT=20
    face=3Darial size=3D2>&nbsp;&nbsp;&nbsp; Helix( RealOne forum) say =
that I would=20
    need </FONT><FONT face=3D"Times New Roman, Times">ISO/IEC 1172 for =
RFC=20
    </FONT><BR><FONT face=3Darial size=3D2>&nbsp;&nbsp;&nbsp; =
implementation. Where=20
    can I get </FONT><FONT face=3D"Times New Roman, Times">ISO/IEC 1172=20
    </FONT><FONT face=3Darial size=3D2>from, any inputs on this one=20
    please</FONT><BR>&nbsp;<BR><FONT face=3Darial size=3D2>Here is the =
extract of=20
    the document from real networks</FONT><BR><FONT face=3Darial=20
    =
size=3D2>***********************</FONT><BR>=B7&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;=20
    MPEG-1<BR>Helix Universal Server delivers ISO/IEC 1172 compliant =
video and=20
    system bit <BR>streams over RTSP/RTP. RealOne Player and QuickTime =
Player,=20
    for example, <BR>can play MPEG-1 streams. Helix Universal Server =
delivers=20
    MPEG-1 content <BR>with file extensions of mpa, mpg, mpeg, mpv, mps, =
m2v,=20
    m1v, and mpe<BR><FONT face=3Darial=20
    size=3D2>***********************</FONT><BR>&nbsp;<BR><FONT =
face=3Darial=20
    size=3D2>Thanks and Regards</FONT><BR><FONT face=3Darial=20
  size=3D2>Deepak</FONT></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

--Boundary_(ID_oxjF1x8snIqivsFUXbmBYw)--

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Wed, 27 Aug 2003 08:28:28 -0700 (PDT)
From: Stephen Casner <casner@acm.org>
To: Oren Peleg <OrenP@audiocodes.com>
cc: Henning Schulzrinne <hgs@cs.columbia.edu>, avt@ietf.org
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Subject: [AVT] RE: Question on 4.1 section of RFC3551
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On Wed, 27 Aug 2003, Oren Peleg wrote:

> Yet, if a CN is present you can distinguish between silence or
> unknown state of the session connection & you can use a CNG to give
> better QoS to the listener.  Your point is well taken - Off course
> the receiver SHOULD be robust, but still, In a Coder point of view,
> (I am not sure this working group is the right address, but still
> ..) there is the G711 Appendix II which defines silence
> compression. Or should I ask, when stopping the packet sending due
> to silence, what is the standard used for doing so?

None other than RFC 3551.

                                                        -- Steve

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Oren Peleg" <OrenP@audiocodes.com>
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Yet, if a CN is present you can distinguish between silence or unknown =
state of the session connection & you can use a CNG to give better QoS =
to the listener.  Your point is well taken - Off course the receiver =
SHOULD be robust, but still, In a Coder point of view, (I am not sure =
this working group is the right address, but still ..) there is the G711 =
Appendix II which defines silence compression. Or should I ask, when =
stopping the packet sending due to silence, what is the standard used =
for doing so?

Thanks,

Oren P.

-----Original Message-----
From: Stephen Casner [mailto:casner@acm.org]
Sent: Tuesday, August 26, 2003 11:27 PM
To: Henning Schulzrinne
Cc: Oren Peleg; avt@ietf.org
Subject: Re: Question on 4.1 section of RFC3551


On Tue, 26 Aug 2003, Henning Schulzrinne wrote:

> For example, in a conference bridge, it would be counterproductive to
> send CN packets for all but the active speaker(s). Also, IETF specs
> aren't product sales and training manuals - it is not the IETF's job =
to
> tell people that engineering systems without CN is generally a bad =
idea.
> Sending CN is not required for interoperability, so being too
> prescriptive doesn't seem helpful here.

I agree with Henning's sentiments here.  To go further, I would say
that the receiver SHOULD be prepared to work well even if CN is not
received, because it is always possible that the CN packet may be lost
even if it was sent.

Many miles were logged on the MBone tools (vat, RAT, NeVoT and
friends) with no CN.  You may choose to discount these as not be of
service grade, but they worked properly without requiring CN.  When we
have multicast sessions from the IETF meeting with hundreds of
participants, we were glad that those hundreds of participants weren't
sending CN all the time.  Yet at any point that any one of them wanted
to say something back to the session, the communication path was
ready.

                                                        -- Steve

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Wed Aug 27 22:31:18 2003
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Date: Wed, 27 Aug 2003 17:55:25 +0100
Subject: Re: [AVT] Interpreting RFC2327: SDP whitespace problem in "s=" line
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To: Thomas Zeng <zeng@PacketVideo.COM>
From: Colin Perkins <csp@csperkins.org>
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On Wednesday, Aug 27, 2003, at 01:29 Europe/London, Thomas Zeng wrote:
> Hello All:
>
> During recent interop testing, one issue has come up regarding whether 
> it is
> valid to have "s=[single space]" in
> an SDP as the subject of a presentation.
>
> In section 6 of RFC2327 (see
> http://www.ietf.org/rfc/rfc2327.txt?number=2327):
>
>    An SDP session description consists of a number of lines of text of
>    the form <type>=<value> <type> is always exactly one character and 
> is
>    case-significant.  <value> is a structured text string whose format
>    depends on <type>.  It also will be case-significant unless a
>    specific field defines otherwise.  Whitespace is not permitted 
> either
>    side of the `=' sign. In general <value> is either a number of 
> fields
>    delimited by a single space character or a free format string.
>
> Taken off context, this paragraph seems to be prohibiting
> '<type>[space]=<value>'  or
> '<type>=[space]<value>' .  However, we don't think the intent was to 
> place
> restrictions on <value>.  If <value> is a free-format string, then it 
> should
> be able to start with a space. (And this is what the ABNF definition 
> says,
> for the value in the 's=' case).
>
> We checked
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-new-13.txt as
> well, which clearly states that
> if there is no meaningful subject, a single space should be used in 
> "s="
> line, yet it apparently has contracdictory statement in section 5.
>
> We hereby would like to confirm with this mailing list, that "s=[single
> space]" is a legal subject line.

I would consider it to be legal. RFC 3264 (offer/answer) also 
recommends either a single space or a single dash ("-") character in 
the s= line.

-- 
Colin Perkins
http://csperkins.org/


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Wed, 27 Aug 2003 21:58:52 +0530
From: Deepak Kotian <dkotian3@vsnl.net>
Subject: Re: Re: Re: [AVT] RTP timestamp calculation for MPEG1 Video and Audio
 ESsothat itworks with RTSP client like RealOne/QuickTime
To: Neo@wiseone.zzn.com, avt@ietf.org
Reply-to: Deepak Kotian <dkotian3@vsnl.net>
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Thanks for the information.

I think, there is no soft copy available for this one. Do you
know where I can buy it online please.

Could please shell out a formula something like audio for Video ES.

The PTS of 33 bits, I meant from docs similar to these.
http://www.cs.columbia.edu/~hgs/rtp/drafts/draft-hoffman-rtp-mpeg-encap-01.t
xt
*****Extract*****
These PES
packets contain presentation time stamps (PTS). The MPEG2 System standard
specifies that the time interval between contiguous PTS's should not be
greater than 0.7 seconds. We propose that PTS's should be present in every
RTP fixed header. In the MPEG2 System standard the PTS is defined to 33
bits with an accuracy of 90 kHz.  However, in this RTP encapsulation we use
only the 32 LSB's. Consequently, the PTS would cycle every 13.26 hours.
*****Extract*****

Also, I have heard that for B picture frame the timestamp needs to reduced
as well.
Could you elaborate on those lines as well.

Thanks very much for the help extended, really appreciated.
Some more help would make my life easier.

Thanks and Regards
Deepak

----- Original Message -----
From: "Neo Anderson" <neo@wiseone.zzn.com>
To: <dkotian3@vsnl.net>; <avt@ietf.org>
Sent: Wednesday, August 27, 2003 6:42 PM
Subject: Re: Re: Re: [AVT] RTP timestamp calculation for MPEG1 Video and
Audio ESsothat itworks with RTSP client like RealOne/QuickTime


> >>...Which is usually 33 bits, that too is mystery to me as
> >>the RTP timestamp is 32 bits.
>
> First, I am not able to relate the 33 bits you have mentioned to any
> timestamp at all (kindly enlighten with reference)!
> and you might have to buy ISO/IEC 11172-2 for reference if u need
> to...
>
> TS calc in a nutshell:
> If u are streaming from a stored file:
>
> Else, some public source decoder will help to get how to extract the
> temporal_reference and the picture_rate/frame_rate_code fields from
> the MPEG-1 ES. From these two parameters, timestamp can be calculated.
>
> When u have the encoder running live at the constant frame rate, the
> timestamp to be carried MAY be directly given by the encoder or we
> can sample the system clock (gettimeofday() etc.) for the RTP
> timestamp (which implies PTS will be sent in the RTP packet) at the
> instant when the frame is generated.
>
> For Audio,
> If it is constant bitrate (which I presume will be in most cases of
> streaming), then you will have to increment the timestamp based on
> the samples per frame. RTP timestamp will be the timestamp for the
> first audio frame in the RTP packet (it is most likely that multiple
> audio frames are carried in the RTP packet). convert this into the
> resolution as required by rfc2250 (90kHz). Also check out the RTP FAQ
> on this.
> Example:
> Sampling Rate = Fs Hz
> Calculated Samples per frame = x
> Timestamp in 90kHz units = ((x/Fs) * 90000)
>
> For variable bit rate audio,
> Calculate the number of samples per frame for each individual audio
> frame. Based on the sampling frequency, calculate the timestamp which
> is to be sent for each frame and use the timestamp of the first audio
> frame in the rtp packet.
>
> To calculate the number of samples, you will find lot of info on the
> web for the MPEG1 audio frame structure and related formula.
> http://www.wotsit.org is one of them. Check for mp3 frame format for
> audio.
>
> >>You guys are my last hope on this, the helix guys have redirected me
> >>to this forum.
> Hopefully now...
>
>
>
> --Neo
> Get your own Free E-mail at Witch Wisdom http://www.witchwisdom.com
> __________________
> http://www.zzn.com



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: "Deepak Kotian" <dkotian3@vsnl.net>, avt@ietf.org
From: "Sandy (Alexander) MacInnis" <macinnis@broadcom.com>
Subject: Re: [AVT] Where can I get ISO/IEC 1172 from
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<html>
The MPEG-1 standard is ISO/IEC 11172 (not 1172). That looks like a
typo.<br>
<br>
The systems part is 11172-1, video is 11172-2, and audio is=20
11172-3.<br>
<br>
These standards are copyrighted by ISO/IEC, and they are available in
hard copy from the national standards bodies and IEC.<br>
<br>
There may be some people with bootleg non-final soft copies, but as far
as I know they are not permitted to be distributed due to=20
copyright.<br>
<br>
--Sandy MacInnis<br>
Broadcom<br>
(Chair of MPEG-1 Systems and MPEG-2 Systems at the time they were
developed.)<br>
<br>
At 02:42 PM 8/26/2003, Deepak Kotian wrote:<br>
<blockquote type=3Dcite cite><font face=3D"arial" size=3D2>Hi,</font><br>
&nbsp;<br>
<font face=3D"arial" size=3D2>&nbsp;&nbsp;&nbsp; I want to implement RFC 225=
0
for MPEG1 Video and Audio ES.</font><br>
<font face=3D"arial" size=3D2>&nbsp;&nbsp;&nbsp; Helix( RealOne forum) say
that I would need </font><font face=3D"Times New Roman, Times">ISO/IEC 1172
for RFC </font><br>
<font face=3D"arial" size=3D2>&nbsp;&nbsp;&nbsp; implementation. Where can I
get </font><font face=3D"Times New Roman, Times">ISO/IEC 1172
</font><font face=3D"arial" size=3D2>from, any inputs on this one
please</font><br>
&nbsp;<br>
<font face=3D"arial" size=3D2>Here is the extract of the document from real
networks</font><br>
<font face=3D"arial" size=3D2>***********************</font><br>
=B7&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; MPEG-1<br>
Helix Universal Server delivers ISO/IEC 1172 compliant video and system
bit <br>
streams over RTSP/RTP. RealOne Player and QuickTime Player, for example,
<br>
can play MPEG-1 streams. Helix Universal Server delivers MPEG-1 content
<br>
with file extensions of mpa, mpg, mpeg, mpv, mps, m2v, m1v, and mpe<br>
<font face=3D"arial" size=3D2>***********************</font><br>
&nbsp;<br>
<font face=3D"arial" size=3D2>Thanks and Regards</font><br>
<font face=3D"arial" size=3D2>Deepak</font></blockquote></html>



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Thu, 28 Aug 2003 00:14:48 +0530
From: Deepak Kotian <dkotian3@vsnl.net>
Subject: Re: [AVT] Where can I get ISO/IEC 1172 from
To: Colin Perkins <csp@csperkins.org>
Cc: "Sandy (Alexander) MacInnis" <macinnis@broadcom.com>, avt@ietf.org
Reply-to: Deepak Kotian <dkotian3@vsnl.net>
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The cost seems to be 44,00 swiss francs, I do not have that
much of budget to shell on this right now.
Unless there is some other way to get it.

I  just need a proper RTP timestamp calcualation for MPEG1 audio
and Video ES considering various frames and I think someone in the gr=
oup
should have some handy docuement on it.
Can someone share it generously.


Thanks and Regards
Deepak

----- Original Message -----
=46rom: "Colin Perkins" <csp@csperkins.org>
To: "Deepak Kotian" <dkotian3@vsnl.net>
Cc: "Sandy (Alexander) MacInnis" <macinnis@broadcom.com>; <avt@ietf.o=
rg>
Sent: Wednesday, August 27, 2003 11:23 PM
Subject: Re: [AVT] Where can I get ISO/IEC 1172 from


I believe you can now buy copies of ISO standards online, available f=
or
download as PDF files. See
http://www.iso.ch/iso/en/prods-services/ISOstore/store.html

Colin


On Wednesday, Aug 27, 2003, at 04:03 Europe/London, Deepak Kotian wro=
te:

> Thanks for the input.
> So, basically it is difficult to get it.
> Can someone in the forum have a handy document for calculation of R=
TP
> timestamp with reference
> to PTS/DTS for Video and Audio ES.
>
> Any help on this would be great.
>
> Thanks and Regards
> Deepak
>
> ----- Original Message -----
> From: Sandy (Alexander) MacInnis
> To: Deepak Kotian ; avt@ietf.org
> Sent: Wednesday, August 27, 2003 6:09 AM
> Subject: Re: [AVT] Where can I get ISO/IEC 1172 from
>
> The MPEG-1 standard is ISO/IEC 11172 (not 1172). That looks like a
> typo.
>
> The systems part is 11172-1, video is 11172-2, and audio is 11172-3=
.
>
> These standards are copyrighted by ISO/IEC, and they are available =
in
> hard copy from the national standards bodies and IEC.
>
> There may be some people with bootleg non-final soft copies, but as
> far as I know they are not permitted to be distributed due to
> copyright.
>
> --Sandy MacInnis
> Broadcom
> (Chair of MPEG-1 Systems and MPEG-2 Systems at the time they were
> developed.)
>
> At 02:42 PM 8/26/2003, Deepak Kotian wrote:
>
> Hi,
>
> I want to implement RFC 2250 for MPEG1 Video and Audio ES.
> Helix( RealOne forum) say that I would need ISO/IEC 1172 for RFC
> implementation. Where can I get ISO/IEC 1172 from, any inputs on
> this one please
>
> Here is the extract of the document from real networks
> ***********************
> =B7 MPEG-1
> Helix Universal Server delivers ISO/IEC 1172 compliant video and
> system bit
> streams over RTSP/RTP. RealOne Player and QuickTime Player, for
> example,
> can play MPEG-1 streams. Helix Universal Server delivers MPEG-1 con=
tent
> with file extensions of mpa, mpg, mpeg, mpv, mps, m2v, m1v, and mpe
> ***********************
>
> Thanks and Regards
> Deepak
>
>
--
Colin Perkins
http://csperkins.org/






_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: "Deepak Kotian" <dkotian3@vsnl.net>, "Colin Perkins" <csp@csperkins.org>
From: "Sandy (Alexander) MacInnis" <macinnis@broadcom.com>
Subject: Re: [AVT] Where can I get ISO/IEC 1172 from
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The MPEG-1 Systems standard doesn't say anything directly about RTP.

Fundamentally, it specifies a formal timing model for determining the 
conformance of bit streams, and this timing model uses time stamps.

There are 3 types of time stamps included:

- SCR (System Clock Reference) - this is in effect a sample of the stream's 
(i.e. sender's) time base as it applies to the compressed stream. To the 
receiver it means, in effect, "When you get this, the time is the value in 
this SCR".

- PTS (Presentation Time Stamp) - indicates the presentation (i.e. display) 
time of a picture. This is always delayed from the time the compressed data 
at the start of a picture arrives, since there is a non-negative delay in 
the compressed data buffer and it takes time to transfer a picture across a 
finite rate channel.

- DTS (Decoding Time Stamp) - indicates the decoding time of a picture, in 
case it is different from the PTS. This is really to easy conformance 
testing, decoders don't need to make use of it. DTS != PTS in the case of I 
and P pictures in streams that may have B pictures, due to the re-ordering 
delay that is inherent in the use of B pictures.

All of these refer to an idealized decoder model called an STD (System 
Target Decoder); there are significant differences between the STD model 
and most real decoders. For example, in the STD, pictures are decoded 
instantaneously, and they are displayed instantaneously (not with raster 
scan). Those differences are not relevant to an encoder - the requirements 
of the stream are set by the standard, not by individual decoders. However 
decoders typically have to make some adjustments to account for the time 
required to decode pictures, etc.

All of the time stamps (SCR, PTS, DTS) are samples of same underlying time 
base. This is fundamental to the way MPEG Systems works (MPEG-1, MPEG-2 
Transport, and MPEG-2 Program streams).

--Sandy

At 11:35 AM 8/27/2003, Deepak Kotian wrote:
>Thanks for the link.
>I assume, it will really help for RTP timestamp calculation
>for realTime video generated from encoder boards.
>
>Thanks and Regards
>Deepak



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Thu, 28 Aug 2003 00:05:32 +0530
From: Deepak Kotian <dkotian3@vsnl.net>
Subject: Re: [AVT] Where can I get ISO/IEC 1172 from
To: Colin Perkins <csp@csperkins.org>
Cc: "Sandy (Alexander) MacInnis" <macinnis@broadcom.com>, avt@ietf.org
Reply-to: Deepak Kotian <dkotian3@vsnl.net>
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Thanks for the link.
I assume, it will really help for RTP timestamp calculation
for realTime video generated from encoder boards.

Thanks and Regards
Deepak

----- Original Message -----
=46rom: "Colin Perkins" <csp@csperkins.org>
To: "Deepak Kotian" <dkotian3@vsnl.net>
Cc: "Sandy (Alexander) MacInnis" <macinnis@broadcom.com>; <avt@ietf.o=
rg>
Sent: Wednesday, August 27, 2003 11:23 PM
Subject: Re: [AVT] Where can I get ISO/IEC 1172 from


I believe you can now buy copies of ISO standards online, available f=
or
download as PDF files. See
http://www.iso.ch/iso/en/prods-services/ISOstore/store.html

Colin


On Wednesday, Aug 27, 2003, at 04:03 Europe/London, Deepak Kotian wro=
te:

> Thanks for the input.
> So, basically it is difficult to get it.
> Can someone in the forum have a handy document for calculation of R=
TP
> timestamp with reference
> to PTS/DTS for Video and Audio ES.
>
> Any help on this would be great.
>
> Thanks and Regards
> Deepak
>
> ----- Original Message -----
> From: Sandy (Alexander) MacInnis
> To: Deepak Kotian ; avt@ietf.org
> Sent: Wednesday, August 27, 2003 6:09 AM
> Subject: Re: [AVT] Where can I get ISO/IEC 1172 from
>
> The MPEG-1 standard is ISO/IEC 11172 (not 1172). That looks like a
> typo.
>
> The systems part is 11172-1, video is 11172-2, and audio is 11172-3=
.
>
> These standards are copyrighted by ISO/IEC, and they are available =
in
> hard copy from the national standards bodies and IEC.
>
> There may be some people with bootleg non-final soft copies, but as
> far as I know they are not permitted to be distributed due to
> copyright.
>
> --Sandy MacInnis
> Broadcom
> (Chair of MPEG-1 Systems and MPEG-2 Systems at the time they were
> developed.)
>
> At 02:42 PM 8/26/2003, Deepak Kotian wrote:
>
> Hi,
>
> I want to implement RFC 2250 for MPEG1 Video and Audio ES.
> Helix( RealOne forum) say that I would need ISO/IEC 1172 for RFC
> implementation. Where can I get ISO/IEC 1172 from, any inputs on
> this one please
>
> Here is the extract of the document from real networks
> ***********************
> =B7 MPEG-1
> Helix Universal Server delivers ISO/IEC 1172 compliant video and
> system bit
> streams over RTSP/RTP. RealOne Player and QuickTime Player, for
> example,
> can play MPEG-1 streams. Helix Universal Server delivers MPEG-1 con=
tent
> with file extensions of mpa, mpg, mpeg, mpv, mps, m2v, m1v, and mpe
> ***********************
>
> Thanks and Regards
> Deepak
>
>
--
Colin Perkins
http://csperkins.org/






_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: Re: [AVT] Where can I get ISO/IEC 1172 from
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Cc: "Sandy (Alexander) MacInnis" <macinnis@broadcom.com>, avt@ietf.org
To: Deepak Kotian <dkotian3@vsnl.net>
From: Colin Perkins <csp@csperkins.org>
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I believe you can now buy copies of ISO standards online, available for=20=

download as PDF files. See=20
http://www.iso.ch/iso/en/prods-services/ISOstore/store.html

Colin


On Wednesday, Aug 27, 2003, at 04:03 Europe/London, Deepak Kotian wrote:

> Thanks for the input.
> So, basically it is difficult to get it.
> Can someone in the forum have a handy document for calculation of RTP=20=

> timestamp with reference
> to PTS/DTS for Video and Audio ES.
> =A0
> Any help on this would be great.
> =A0
> Thanks and Regards
> Deepak
>
> ----- Original Message -----
> From: Sandy (Alexander) MacInnis
> To: Deepak Kotian ; avt@ietf.org
> Sent: Wednesday, August 27, 2003 6:09 AM
> Subject: Re: [AVT] Where can I get ISO/IEC 1172 from
>
> The MPEG-1 standard is ISO/IEC 11172 (not 1172). That looks like a=20
> typo.
>
> The systems part is 11172-1, video is 11172-2, and audio is 11172-3.
>
> These standards are copyrighted by ISO/IEC, and they are available in=20=

> hard copy from the national standards bodies and IEC.
>
> There may be some people with bootleg non-final soft copies, but as=20
> far as I know they are not permitted to be distributed due to=20
> copyright.
>
> --Sandy MacInnis
> Broadcom
> (Chair of MPEG-1 Systems and MPEG-2 Systems at the time they were=20
> developed.)
>
> At 02:42 PM 8/26/2003, Deepak Kotian wrote:
>
> Hi,
> =A0
> =A0=A0=A0 I want to implement RFC 2250 for MPEG1 Video and Audio ES.
> =A0=A0=A0 Helix( RealOne forum) say that I would need ISO/IEC 1172 for =
RFC
> =A0=A0=A0 implementation. Where can I get ISO/IEC 1172 from, any =
inputs on=20
> this one please
> =A0
> Here is the extract of the document from real networks
> ***********************
> =B7=A0=A0=A0=A0=A0=A0 MPEG-1
> Helix Universal Server delivers ISO/IEC 1172 compliant video and=20
> system bit
> streams over RTSP/RTP. RealOne Player and QuickTime Player, for=20
> example,
> can play MPEG-1 streams. Helix Universal Server delivers MPEG-1 =
content
> with file extensions of mpa, mpg, mpeg, mpv, mps, m2v, m1v, and mpe
> ***********************
> =A0
> Thanks and Regards
> Deepak
>
>
--=20
Colin Perkins
http://csperkins.org/


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] Where can I get ISO/IEC 1172 from
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Thread-Topic: [AVT] Where can I get ISO/IEC 1172 from
thread-index: AcNs6Cq3z7g2WdupRdqu8lUX4bnWnAAB1TfA
From: "Gary Sullivan" <garysull@windows.microsoft.com>
To: "Sandy \(Alexander\) MacInnis" <macinnis@broadcom.com>,
        "Deepak Kotian" <dkotian3@vsnl.net>, <avt@ietf.org>
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Fully-legit electronic copies may be purchased from the following link:
=20
http://tinyurl.com/lewc=20
=20
(It's not even expensive -- about $31 for either video or systems.)=20
=20
-Gary Sullivan
=20

-----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On Behalf Of Sandy
(Alexander) MacInnis
Sent: Tuesday, August 26, 2003 5:39 PM
To: Deepak Kotian; avt@ietf.org
Subject: Re: [AVT] Where can I get ISO/IEC 1172 from


The MPEG-1 standard is ISO/IEC 11172 (not 1172). That looks like a typo.

The systems part is 11172-1, video is 11172-2, and audio is 11172-3.

These standards are copyrighted by ISO/IEC, and they are available in
hard copy from the national standards bodies and IEC.

There may be some people with bootleg non-final soft copies, but as far
as I know they are not permitted to be distributed due to copyright.

--Sandy MacInnis
Broadcom
(Chair of MPEG-1 Systems and MPEG-2 Systems at the time they were
developed.)

At 02:42 PM 8/26/2003, Deepak Kotian wrote:


Hi,
=20
    I want to implement RFC 2250 for MPEG1 Video and Audio ES.
    Helix( RealOne forum) say that I would need ISO/IEC 1172 for RFC=20
    implementation. Where can I get ISO/IEC 1172 from, any inputs on
this one please
=20
Here is the extract of the document from real networks
***********************
*       MPEG-1
Helix Universal Server delivers ISO/IEC 1172 compliant video and system
bit=20
streams over RTSP/RTP. RealOne Player and QuickTime Player, for example,

can play MPEG-1 streams. Helix Universal Server delivers MPEG-1 content=20
with file extensions of mpa, mpg, mpeg, mpv, mps, m2v, m1v, and mpe
***********************
=20
Thanks and Regards
Deepak

_______________________________________________ Audio/Video Transport
Working Group avt@ietf.org https://www1.ietf.org/mailman/listinfo/avt=20


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<BODY>
<DIV><SPAN class=3D019590323-27082003><FONT face=3D"Courier New"=20
color=3D#0000ff>Fully-legit electronic copies may be purchased from the =
following=20
link:</FONT></SPAN></DIV>
<DIV><SPAN class=3D019590323-27082003><FONT face=3D"Courier New"=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D019590323-27082003><FONT face=3D"Courier New" =
color=3D#0000ff><A=20
href=3D"http://tinyurl.com/lewc">http://tinyurl.com/lewc</A> =
</FONT></SPAN></DIV>
<DIV><SPAN class=3D019590323-27082003><FONT=20
face=3D"Courier New"></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D019590323-27082003><FONT face=3D"Courier New">(It's =
not even=20
expensive -- about $31 for either video or =
systems.)&nbsp;</FONT></SPAN></DIV>
<DIV><SPAN class=3D019590323-27082003><FONT face=3D"Courier New"=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D019590323-27082003><FONT face=3D"Courier New" =
color=3D#0000ff>-Gary=20
Sullivan</FONT></SPAN></DIV>
<DIV><SPAN class=3D019590323-27082003><FONT face=3D"Courier New"=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B>=20
  avt-admin@ietf.org [mailto:avt-admin@ietf.org] <B>On Behalf Of =
</B>Sandy=20
  (Alexander) MacInnis<BR><B>Sent:</B> Tuesday, August 26, 2003 5:39=20
  PM<BR><B>To:</B> Deepak Kotian; avt@ietf.org<BR><B>Subject:</B> Re: =
[AVT]=20
  Where can I get ISO/IEC 1172 from<BR><BR></FONT></DIV>The MPEG-1 =
standard is=20
  ISO/IEC 11172 (not 1172). That looks like a typo.<BR><BR>The systems =
part is=20
  11172-1, video is 11172-2, and audio is 11172-3.<BR><BR>These =
standards are=20
  copyrighted by ISO/IEC, and they are available in hard copy from the =
national=20
  standards bodies and IEC.<BR><BR>There may be some people with bootleg =

  non-final soft copies, but as far as I know they are not permitted to =
be=20
  distributed due to copyright.<BR><BR>--Sandy =
MacInnis<BR>Broadcom<BR>(Chair of=20
  MPEG-1 Systems and MPEG-2 Systems at the time they were =
developed.)<BR><BR>At=20
  02:42 PM 8/26/2003, Deepak Kotian wrote:<BR>
  <BLOCKQUOTE cite=3D"" type=3D"cite"><FONT face=3Darial=20
    size=3D2>Hi,</FONT><BR>&nbsp;<BR><FONT face=3Darial =
size=3D2>&nbsp;&nbsp;&nbsp; I=20
    want to implement RFC 2250 for MPEG1 Video and Audio =
ES.</FONT><BR><FONT=20
    face=3Darial size=3D2>&nbsp;&nbsp;&nbsp; Helix( RealOne forum) say =
that I would=20
    need </FONT><FONT face=3D"Times New Roman, Times">ISO/IEC 1172 for =
RFC=20
    </FONT><BR><FONT face=3Darial size=3D2>&nbsp;&nbsp;&nbsp; =
implementation. Where=20
    can I get </FONT><FONT face=3D"Times New Roman, Times">ISO/IEC 1172=20
    </FONT><FONT face=3Darial size=3D2>from, any inputs on this one=20
    please</FONT><BR>&nbsp;<BR><FONT face=3Darial size=3D2>Here is the =
extract of=20
    the document from real networks</FONT><BR><FONT face=3Darial=20
    =
size=3D2>***********************</FONT><BR>&middot;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
    MPEG-1<BR>Helix Universal Server delivers ISO/IEC 1172 compliant =
video and=20
    system bit <BR>streams over RTSP/RTP. RealOne Player and QuickTime =
Player,=20
    for example, <BR>can play MPEG-1 streams. Helix Universal Server =
delivers=20
    MPEG-1 content <BR>with file extensions of mpa, mpg, mpeg, mpv, mps, =
m2v,=20
    m1v, and mpe<BR><FONT face=3Darial=20
    size=3D2>***********************</FONT><BR>&nbsp;<BR><FONT =
face=3Darial=20
    size=3D2>Thanks and Regards</FONT><BR><FONT face=3Darial=20
  =
size=3D2>Deepak</FONT></BLOCKQUOTE>______________________________________=
_________=20
  Audio/Video Transport Working Group avt@ietf.org=20
  https://www1.ietf.org/mailman/listinfo/avt </BLOCKQUOTE></BODY></HTML>

------_=_NextPart_001_01C36CF3.102CBB52--

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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From exim@www1.ietf.org  Thu Aug 28 09:36:34 2003
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Date: Thu, 28 Aug 2003 08:20:24 +0530
From: Deepak Kotian <dkotian3@vsnl.net>
Subject: Re: [AVT] Where can I get ISO/IEC 1172 from
To: "Sandy (Alexander) MacInnis" <macinnis@broadcom.com>,
        Colin Perkins <csp@csperkins.org>
Cc: avt@ietf.org
Reply-to: Deepak Kotian <dkotian3@vsnl.net>
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 Thanks for inputs.
Just telling you the backgorund.
I have an encoder which sends out Video ES and Audio ES, I want
to send it out in RTP format to play it on RealOne player. We have
supported RTSP Minimal server and RealOne player plays Video with some
breaks
and latency and 2 sec audio after 1 minute. Do not know the exact reason.
So, we are doubting the RTP timestamp calculation which is not
clear from the RFC. We have done encapsulation as per RFC 2250.

Any suggestions on this


List my query/problem in a simpler way.

1.  Where do I source the Time to be fed in RTP timestamp field in the
      RTP header in my case, should it be PTS/DTS/SCR. This is mainly for
      MPEG1 for Video ES
      Is there some calculation needed for the same.

2.   Also, If the calculation is to be done from MPEG1 headers, the SCR/PTS
field
      seems to of 33 bits, how do we fit it in 32 bit timestamp of RTP.

3.  Should the SCR or PTS or DTS be multiplied by 90 and fed in the RTP
timestamp.

4.  Is there a special way to calculate time for B-Frame for Video ES.


Can one over here kindly give their  view on these queries.

Thanks and Regards
Deepak





----- Original Message -----
From: "Sandy (Alexander) MacInnis" <macinnis@broadcom.com>
To: "Deepak Kotian" <dkotian3@vsnl.net>; "Colin Perkins" <csp@csperkins.org>
Cc: <avt@ietf.org>
Sent: Thursday, August 28, 2003 12:31 AM
Subject: Re: [AVT] Where can I get ISO/IEC 1172 from


> The MPEG-1 Systems standard doesn't say anything directly about RTP.
>
> Fundamentally, it specifies a formal timing model for determining the
> conformance of bit streams, and this timing model uses time stamps.
>
> There are 3 types of time stamps included:
>
> - SCR (System Clock Reference) - this is in effect a sample of the
stream's
> (i.e. sender's) time base as it applies to the compressed stream. To the
> receiver it means, in effect, "When you get this, the time is the value in
> this SCR".
>
> - PTS (Presentation Time Stamp) - indicates the presentation (i.e.
display)
> time of a picture. This is always delayed from the time the compressed
data
> at the start of a picture arrives, since there is a non-negative delay in
> the compressed data buffer and it takes time to transfer a picture across
a
> finite rate channel.
>
> - DTS (Decoding Time Stamp) - indicates the decoding time of a picture, in
> case it is different from the PTS. This is really to easy conformance
> testing, decoders don't need to make use of it. DTS != PTS in the case of
I
> and P pictures in streams that may have B pictures, due to the re-ordering
> delay that is inherent in the use of B pictures.
>
> All of these refer to an idealized decoder model called an STD (System
> Target Decoder); there are significant differences between the STD model
> and most real decoders. For example, in the STD, pictures are decoded
> instantaneously, and they are displayed instantaneously (not with raster
> scan). Those differences are not relevant to an encoder - the requirements
> of the stream are set by the standard, not by individual decoders. However
> decoders typically have to make some adjustments to account for the time
> required to decode pictures, etc.
>
> All of the time stamps (SCR, PTS, DTS) are samples of same underlying time
> base. This is fundamental to the way MPEG Systems works (MPEG-1, MPEG-2
> Transport, and MPEG-2 Program streams).
>
> --Sandy
>
> At 11:35 AM 8/27/2003, Deepak Kotian wrote:
> >Thanks for the link.
> >I assume, it will really help for RTP timestamp calculation
> >for realTime video generated from encoder boards.
> >
> >Thanks and Regards
> >Deepak
>
>



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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comments inline

On Thu, 28 Aug 2003, Deepak Kotian wrote:

>  Thanks for inputs.
> Just telling you the backgorund.
> I have an encoder which sends out Video ES and Audio ES, I want
> to send it out in RTP format to play it on RealOne player. We have
> supported RTSP Minimal server and RealOne player plays Video with some
> breaks
> and latency and 2 sec audio after 1 minute. Do not know the exact reason.
> So, we are doubting the RTP timestamp calculation which is not
> clear from the RFC. We have done encapsulation as per RFC 2250.
> 
> Any suggestions on this
> 
> 
> List my query/problem in a simpler way.
> 
> 1.  Where do I source the Time to be fed in RTP timestamp field in the
>       RTP header in my case, should it be PTS/DTS/SCR. This is mainly for
>       MPEG1 for Video ES
>       Is there some calculation needed for the same.

If you look at the RTP RFCs (RFC 1889 or RFC 3550) you will see that it 
says "The timestamp reflects the sampling instant of the first octet in 
the RTP data packet". This statement implies PTS. Later in that section it 
specifically talks about MPEG. It says 
  "Consecutive RTP packets MAY contain timestamps that are not monotonic 
   if the data is not transmitted in the order it was sampled, as in the 
   case of MPEG interpolated video frames."

This again implies PTS because in the case where B frames are used the PTS 
timestamps are not monotonically increasing when the frames are 
transmitted in decode order.

> 
> 2.   Also, If the calculation is to be done from MPEG1 headers, the SCR/PTS
> field
>       seems to of 33 bits, how do we fit it in 32 bit timestamp of RTP.
Drop the 33rd bit. Doing this will just cause the RTP timestamp to wrap 
faster than the MPEG timestamp.

> 
> 3.  Should the SCR or PTS or DTS be multiplied by 90 and fed in the RTP
> timestamp.
I'm not 100% sure about this, but I believe that the PTS sample rate is 
90kHz. That is the same sample rate that the RTP timestamps should be in 
for MPEG. You should not need to modify the timestamps in any way other 
than adding the random offset specified in the RTP spec.

> 
> 4.  Is there a special way to calculate time for B-Frame for Video ES.
See my answer for 1. Just use the PTS timestamps.


Aaron

> 
> 
> Can one over here kindly give their  view on these queries.
> 
> Thanks and Regards
> Deepak
> 
> 
> 
> 
> 
> ----- Original Message -----
> From: "Sandy (Alexander) MacInnis" <macinnis@broadcom.com>
> To: "Deepak Kotian" <dkotian3@vsnl.net>; "Colin Perkins" <csp@csperkins.org>
> Cc: <avt@ietf.org>
> Sent: Thursday, August 28, 2003 12:31 AM
> Subject: Re: [AVT] Where can I get ISO/IEC 1172 from
> 
> 
> > The MPEG-1 Systems standard doesn't say anything directly about RTP.
> >
> > Fundamentally, it specifies a formal timing model for determining the
> > conformance of bit streams, and this timing model uses time stamps.
> >
> > There are 3 types of time stamps included:
> >
> > - SCR (System Clock Reference) - this is in effect a sample of the
> stream's
> > (i.e. sender's) time base as it applies to the compressed stream. To the
> > receiver it means, in effect, "When you get this, the time is the value in
> > this SCR".
> >
> > - PTS (Presentation Time Stamp) - indicates the presentation (i.e.
> display)
> > time of a picture. This is always delayed from the time the compressed
> data
> > at the start of a picture arrives, since there is a non-negative delay in
> > the compressed data buffer and it takes time to transfer a picture across
> a
> > finite rate channel.
> >
> > - DTS (Decoding Time Stamp) - indicates the decoding time of a picture, in
> > case it is different from the PTS. This is really to easy conformance
> > testing, decoders don't need to make use of it. DTS != PTS in the case of
> I
> > and P pictures in streams that may have B pictures, due to the re-ordering
> > delay that is inherent in the use of B pictures.
> >
> > All of these refer to an idealized decoder model called an STD (System
> > Target Decoder); there are significant differences between the STD model
> > and most real decoders. For example, in the STD, pictures are decoded
> > instantaneously, and they are displayed instantaneously (not with raster
> > scan). Those differences are not relevant to an encoder - the requirements
> > of the stream are set by the standard, not by individual decoders. However
> > decoders typically have to make some adjustments to account for the time
> > required to decode pictures, etc.
> >
> > All of the time stamps (SCR, PTS, DTS) are samples of same underlying time
> > base. This is fundamental to the way MPEG Systems works (MPEG-1, MPEG-2
> > Transport, and MPEG-2 Program streams).
> >
> > --Sandy
> >
> > At 11:35 AM 8/27/2003, Deepak Kotian wrote:
> > >Thanks for the link.
> > >I assume, it will really help for RTP timestamp calculation
> > >for realTime video generated from encoder boards.
> > >
> > >Thanks and Regards
> > >Deepak
> >
> >
> 
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> 


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Fri, 29 Aug 2003 00:24:55 +0530
From: Deepak Kotian <dkotian3@vsnl.net>
Subject: Re: [AVT] Where can I get ISO/IEC 1172 from
To: Aaron Colwell <acolwell@real.com>
Cc: "Sandy (Alexander) MacInnis" <macinnis@broadcom.com>,
        Colin Perkins <csp@csperkins.org>, avt@ietf.org
Reply-to: Deepak Kotian <dkotian3@vsnl.net>
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Thanks very much for an excellent response.
We will check this one out and see how it works for us.

1. Also, the RFC 2250 does not say much about how to clearly
encapsulate the MPEG1 Audio ES, frag_offset, frame sync and
timestamp for the same.
Can one give more details on it.
I calculated the as per one of the post, the loading in RealOne
seems faster, but there seems to be audio being played for the
same.

2. Which RFC/document would be good reference for details of MPEG1 system
and
    audio/video ES header details and how MPEG1 packetization
    is done,etc.

Could one please let me know.

Thanks and  Regards
Deepak

----- Original Message -----
From: "Aaron Colwell" <acolwell@real.com>
To: "Deepak Kotian" <dkotian3@vsnl.net>
Cc: "Sandy (Alexander) MacInnis" <macinnis@broadcom.com>; "Colin Perkins"
<csp@csperkins.org>; <avt@ietf.org>
Sent: Thursday, August 28, 2003 9:59 PM
Subject: Re: [AVT] Where can I get ISO/IEC 1172 from


> comments inline
>
> On Thu, 28 Aug 2003, Deepak Kotian wrote:
>
> >  Thanks for inputs.
> > Just telling you the backgorund.
> > I have an encoder which sends out Video ES and Audio ES, I want
> > to send it out in RTP format to play it on RealOne player. We have
> > supported RTSP Minimal server and RealOne player plays Video with some
> > breaks
> > and latency and 2 sec audio after 1 minute. Do not know the exact
reason.
> > So, we are doubting the RTP timestamp calculation which is not
> > clear from the RFC. We have done encapsulation as per RFC 2250.
> >
> > Any suggestions on this
> >
> >
> > List my query/problem in a simpler way.
> >
> > 1.  Where do I source the Time to be fed in RTP timestamp field in the
> >       RTP header in my case, should it be PTS/DTS/SCR. This is mainly
for
> >       MPEG1 for Video ES
> >       Is there some calculation needed for the same.
>
> If you look at the RTP RFCs (RFC 1889 or RFC 3550) you will see that it
> says "The timestamp reflects the sampling instant of the first octet in
> the RTP data packet". This statement implies PTS. Later in that section it
> specifically talks about MPEG. It says
>   "Consecutive RTP packets MAY contain timestamps that are not monotonic
>    if the data is not transmitted in the order it was sampled, as in the
>    case of MPEG interpolated video frames."
>
> This again implies PTS because in the case where B frames are used the PTS
> timestamps are not monotonically increasing when the frames are
> transmitted in decode order.
>
> >
> > 2.   Also, If the calculation is to be done from MPEG1 headers, the
SCR/PTS
> > field
> >       seems to of 33 bits, how do we fit it in 32 bit timestamp of RTP.
> Drop the 33rd bit. Doing this will just cause the RTP timestamp to wrap
> faster than the MPEG timestamp.
>
> >
> > 3.  Should the SCR or PTS or DTS be multiplied by 90 and fed in the RTP
> > timestamp.
> I'm not 100% sure about this, but I believe that the PTS sample rate is
> 90kHz. That is the same sample rate that the RTP timestamps should be in
> for MPEG. You should not need to modify the timestamps in any way other
> than adding the random offset specified in the RTP spec.
>
> >
> > 4.  Is there a special way to calculate time for B-Frame for Video ES.
> See my answer for 1. Just use the PTS timestamps.
>
>
> Aaron
>
> >
> >
> > Can one over here kindly give their  view on these queries.
> >
> > Thanks and Regards
> > Deepak
> >
> >
> >
> >
> >
> > ----- Original Message -----
> > From: "Sandy (Alexander) MacInnis" <macinnis@broadcom.com>
> > To: "Deepak Kotian" <dkotian3@vsnl.net>; "Colin Perkins"
<csp@csperkins.org>
> > Cc: <avt@ietf.org>
> > Sent: Thursday, August 28, 2003 12:31 AM
> > Subject: Re: [AVT] Where can I get ISO/IEC 1172 from
> >
> >
> > > The MPEG-1 Systems standard doesn't say anything directly about RTP.
> > >
> > > Fundamentally, it specifies a formal timing model for determining the
> > > conformance of bit streams, and this timing model uses time stamps.
> > >
> > > There are 3 types of time stamps included:
> > >
> > > - SCR (System Clock Reference) - this is in effect a sample of the
> > stream's
> > > (i.e. sender's) time base as it applies to the compressed stream. To
the
> > > receiver it means, in effect, "When you get this, the time is the
value in
> > > this SCR".
> > >
> > > - PTS (Presentation Time Stamp) - indicates the presentation (i.e.
> > display)
> > > time of a picture. This is always delayed from the time the compressed
> > data
> > > at the start of a picture arrives, since there is a non-negative delay
in
> > > the compressed data buffer and it takes time to transfer a picture
across
> > a
> > > finite rate channel.
> > >
> > > - DTS (Decoding Time Stamp) - indicates the decoding time of a
picture, in
> > > case it is different from the PTS. This is really to easy conformance
> > > testing, decoders don't need to make use of it. DTS != PTS in the case
of
> > I
> > > and P pictures in streams that may have B pictures, due to the
re-ordering
> > > delay that is inherent in the use of B pictures.
> > >
> > > All of these refer to an idealized decoder model called an STD (System
> > > Target Decoder); there are significant differences between the STD
model
> > > and most real decoders. For example, in the STD, pictures are decoded
> > > instantaneously, and they are displayed instantaneously (not with
raster
> > > scan). Those differences are not relevant to an encoder - the
requirements
> > > of the stream are set by the standard, not by individual decoders.
However
> > > decoders typically have to make some adjustments to account for the
time
> > > required to decode pictures, etc.
> > >
> > > All of the time stamps (SCR, PTS, DTS) are samples of same underlying
time
> > > base. This is fundamental to the way MPEG Systems works (MPEG-1,
MPEG-2
> > > Transport, and MPEG-2 Program streams).
> > >
> > > --Sandy
> > >
> > > At 11:35 AM 8/27/2003, Deepak Kotian wrote:
> > > >Thanks for the link.
> > > >I assume, it will really help for RTP timestamp calculation
> > > >for realTime video generated from encoder boards.
> > > >
> > > >Thanks and Regards
> > > >Deepak
> > >
> > >
> >
> >
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www1.ietf.org/mailman/listinfo/avt
> >
>



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Sat, 30 Aug 2003 01:28:58 +0530
From: Deepak Kotian <dkotian3@vsnl.net>
Subject: Re: Re: Re: [AVT] RTP timestamp calculation for MPEG1 Video and Audio
 ESsothat itworks with RTSP client like RealOne/QuickTime
To: Neo@wiseone.zzn.com, avt@ietf.org
Reply-to: Deepak Kotian <dkotian3@vsnl.net>
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About the audio encapsulation part.

These are points I am not clear.

1. I am encapsulating the audio frames based on Audio Frame Sync code.
    I determine the end of the frame if I get a next frame sync code.
    So, one RTP packet consists of one frame.
    Is this correct ?

2.  About the formula
> Sampling Rate = Fs Hz
> Calculated Samples per frame = x
> Timestamp in 90kHz units = ((x/Fs) * 90000)
How do I calculate samples per frame, Could you please stick the extract
over here. I tried the http://www.wotsit.org/search.asp?page=5&s=music
but it says .
If I check mpeghdr.htm file,
It says
******
How to calculate frame size

Read the BitRate, SampleRate and Padding (as value of one or zero) of the
frame header and use the formula:

                                                  FrameSize = 144 * BitRate
/ SampleRate + Padding

Example: BitRate = 128000, SampleRate=441000, Padding=0  ==>  FrameSize=417
bytes
******
But, I am not sure from where 144 has come from. And where does the padding
bit come from.
Moreever, it seems the decimal value is truncated.
Could not find samples per frame calulation.


Could one please let elaborate more on this.
Now we seem to be OK with video and we just need  help on audio.
Please let us know

Thanks and Regards
Deepak

----- Original Message -----
From: "Neo Anderson" <neo@wiseone.zzn.com>
To: <dkotian3@vsnl.net>; <avt@ietf.org>
Sent: Wednesday, August 27, 2003 6:42 PM
Subject: Re: Re: Re: [AVT] RTP timestamp calculation for MPEG1 Video and
Audio ESsothat itworks with RTSP client like RealOne/QuickTime


> >>...Which is usually 33 bits, that too is mystery to me as
> >>the RTP timestamp is 32 bits.
>
> First, I am not able to relate the 33 bits you have mentioned to any
> timestamp at all (kindly enlighten with reference)!
> and you might have to buy ISO/IEC 11172-2 for reference if u need
> to...
>
> TS calc in a nutshell:
> If u are streaming from a stored file:
>
> Else, some public source decoder will help to get how to extract the
> temporal_reference and the picture_rate/frame_rate_code fields from
> the MPEG-1 ES. From these two parameters, timestamp can be calculated.
>
> When u have the encoder running live at the constant frame rate, the
> timestamp to be carried MAY be directly given by the encoder or we
> can sample the system clock (gettimeofday() etc.) for the RTP
> timestamp (which implies PTS will be sent in the RTP packet) at the
> instant when the frame is generated.
>
> For Audio,
> If it is constant bitrate (which I presume will be in most cases of
> streaming), then you will have to increment the timestamp based on
> the samples per frame. RTP timestamp will be the timestamp for the
> first audio frame in the RTP packet (it is most likely that multiple
> audio frames are carried in the RTP packet). convert this into the
> resolution as required by rfc2250 (90kHz). Also check out the RTP FAQ
> on this.
> Example:
> Sampling Rate = Fs Hz
> Calculated Samples per frame = x
> Timestamp in 90kHz units = ((x/Fs) * 90000)
>
> For variable bit rate audio,
> Calculate the number of samples per frame for each individual audio
> frame. Based on the sampling frequency, calculate the timestamp which
> is to be sent for each frame and use the timestamp of the first audio
> frame in the rtp packet.
>
> To calculate the number of samples, you will find lot of info on the
> web for the MPEG1 audio frame structure and related formula.
> http://www.wotsit.org is one of them. Check for mp3 frame format for
> audio.
>
> >>You guys are my last hope on this, the helix guys have redirected me
> >>to this forum.
> Hopefully now...
>
>
>
> --Neo
> Get your own Free E-mail at Witch Wisdom http://www.witchwisdom.com
> __________________
> http://www.zzn.com
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: Re: Re: Re: Re: [AVT] RTP timestamp calculation for MPEG1 Video and AudioESsothat itworks with RTSP client like RealOne/QuickTime
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Comments inline:
>1. I am encapsulating the audio frames based on Audio Frame Sync 
>code.
>   I determine the end of the frame if I get a next frame sync code.
>   So, one RTP packet consists of one frame.
>   Is this correct ?
In cases when there is no intermediate data between the mpeg frames, 
this is correct. (For most practical cases this should hold good).
Else, the size of the frame has to be calculated using the same 
formula as in (2).

(If you are doing live encoding, are you not able to get the frame 
length from the encoder itself?)

>2.  About the formula
>> Sampling Rate = Fs Hz
>> Calculated Samples per frame = x
>> Timestamp in 90kHz units = ((x/Fs) * 90000)
>How do I calculate samples per frame, Could you please stick the 
>extract
>over here. I tried the http://www.wotsit.org/search.asp?
>page=5&s=music

There are some more files. Kindly have a look into them too. (Search 
for mp3 in wotsit)

>but it says .
>If I check mpeghdr.htm file,
>It says
>******
>How to calculate frame size
>
>Read the BitRate, SampleRate and Padding (as value of one or zero) 
of the
>frame header and use the formula:

>FrameSize = 144 * BitRate/ SampleRate + Padding

>Example: BitRate = 128000, SampleRate=441000, Padding=0  ==>  
>FrameSize=417
>bytes
******
>But, I am not sure from where 144 has come from. And where does the 
>padding
>bit come from.

Note 1: SampleRate=441000 should be 44100.

All the bits come from the MPEG frame header which resides after the 
sync. You will have to parse the related info from the frame.

Another good reading of mpeghdr.htm should solve your problem of 
parsing the frame header. 

144 comes like this:
There are 1152 "RAW" samples per frame in layer II & III.
After compression, frame length is calculated in slots.
One slot is 1 byte long for layer II and layer III.
Consider Fs = 44100 and BR=128000

Duration of audio in a frame = 1152/44100 (seconds)

Since we get 128000 bits in a second:
Number of bits in this duration = ((1152/44100) * 128000) * (1), 
where 1 is the number of slots
or 
(((1152/44100)*128000)/8) * (1) BYTES, where 1 is the number of slots.

The variables here are Fs and BR. The constant field is (1152/8) = 
144.

Extract from the same url above:
"Padding is used to fit the bit rates exactly. For an example: 128k 
44.1kHz layer II uses a lot of 418 bytes and some of 417 bytes long 
frames to get the exact 128k bitrate. For Layer I slot is 32 bits 
long, for Layer II and Layer III slot is 8 bits long. "

Suggests you will have a few frames which are padded and a few which 
do not, in order to maintain proper bitrate because we end up in a 
fraction with "0.95" slots left over per frame (0.4 bits i.e. we may 
insert one padded frame for every 19 frames, say and frame length 
will be 418)!

Try other sampling frequencies and you will not end up with a 
fractional value.
44100 and 22050 is maintained for some historical reason to do with 
cd's and tv's...





--Neo
Get your own Free E-mail at Witch Wisdom http://www.witchwisdom.com
__________________
http://www.zzn.com

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: neo@wiseone.zzn.com
Subject: Re: Re: Re: Re: [AVT] RTP timestamp calculation for MPEG1 Video
    and AudioESsothat itworks with RTSP client like RealOne/QuickTime
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Thanks very much for a excellent response.
This is good information.

Also observed :
We just tried sending the PTS from MPEG1 header as it is in
the RTP timestamp field for each MPEG1 packet for audio and it=20
also worked fine and smooth on RealOne.


Thanks and Regards
Deepak




Thanks and Regards
Deepak
neo@wiseone.zzn.com wrote
Comments inline:
>1. I am encapsulating the audio frames based on Audio Frame Sync=20
>code.
>   I determine the end of the frame if I get a next frame sync code.
>   So, one RTP packet consists of one frame.
>   Is this correct ?
In cases when there is no intermediate data between the mpeg frames,=20
this is correct. (For most practical cases this should hold good).
Else, the size of the frame has to be calculated using the same=20
formula as in (2).

(If you are doing live encoding, are you not able to get the frame=20
length from the encoder itself?)

>2.  About the formula
>> Sampling Rate =3D Fs Hz
>> Calculated Samples per frame =3D x
>> Timestamp in 90kHz units =3D ((x/Fs) * 90000)
>How do I calculate samples per frame, Could you please stick the=20
>extract
>over here. I tried the http://www.wotsit.org/search.asp?
>page=3D5&s=3Dmusic

There are some more files. Kindly have a look into them too. (Search=20
for mp3 in wotsit)

>but it says .
>If I check mpeghdr.htm file,
>It says
>******
>How to calculate frame size
>
>Read the BitRate, SampleRate and Padding (as value of one or zero)=20
of the
>frame header and use the formula:

>FrameSize =3D 144 * BitRate/ SampleRate + Padding

>Example: BitRate =3D 128000, SampleRate=3D441000, Padding=3D0  =3D=3D> =20
>FrameSize=3D417
>bytes
******
>But, I am not sure from where 144 has come from. And where does the=20
>padding
>bit come from.

Note 1: SampleRate=3D441000 should be 44100.

All the bits come from the MPEG frame header which resides after the=20
sync. You will have to parse the related info from the frame.

Another good reading of mpeghdr.htm should solve your problem of=20
parsing the frame header.=20

144 comes like this:
There are 1152 "RAW" samples per frame in layer II & III.
After compression, frame length is calculated in slots.
One slot is 1 byte long for layer II and layer III.
Consider Fs =3D 44100 and BR=3D128000

Duration of audio in a frame =3D 1152/44100 (seconds)

Since we get 128000 bits in a second:
Number of bits in this duration =3D ((1152/44100) * 128000) * (1),=20
where 1 is the number of slots
or=20
(((1152/44100)*128000)/8) * (1) BYTES, where 1 is the number of slots.

The variables here are Fs and BR. The constant field is (1152/8) =3D=20
144.

Extract from the same url above:
"Padding is used to fit the bit rates exactly. For an example: 128k=20
44.1kHz layer II uses a lot of 418 bytes and some of 417 bytes long=20
frames to get the exact 128k bitrate. For Layer I slot is 32 bits=20
long, for Layer II and Layer III slot is 8 bits long. "

Suggests you will have a few frames which are padded and a few which=20
do not, in order to maintain proper bitrate because we end up in a=20
fraction with "0.95" slots left over per frame (0.4 bits i.e. we may=20
insert one padded frame for every 19 frames, say and frame length=20
will be 418)!

Try other sampling frequencies and you will not end up with a=20
fractional value.
44100 and 22050 is maintained for some historical reason to do with=20
cd's and tv's...





--Neo
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