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Subject: [rmcat] I-D Action: draft-ietf-rmcat-wireless-tests-02.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the RTP Media Congestion Avoidance Techniques of the IETF.

        Title           : Evaluation Test Cases for Interactive Real-Time Media over Wireless Networks
        Authors         : Zaheduzzaman Sarker
                          Ingemar Johansson
                          Xiaoqing Zhu
                          Jiantao Fu
                          Wei-Tian Tan
                          Michael A. Ramalho
	Filename        : draft-ietf-rmcat-wireless-tests-02.txt
	Pages           : 21
	Date            : 2016-05-07

Abstract:
   It is evident that to ensure seamless and robust user experience
   across all type of access networks multimedia communication suits
   should adapt to the changing network conditions.  There is an ongoing
   effort in IETF RMCAT working group to standardize rate adaptive
   algorithm(s) to be used in the real-time interactive communication.
   In this document test cases are described to evaluate the
   performances of the proposed endpoint adaptation solutions in LTE
   networks and Wi-Fi networks.  The proposed algorithms should be
   evaluated using the test cases defined in this document to select
   most optimal solutions.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rmcat-wireless-tests/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-rmcat-wireless-tests-02

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-rmcat-wireless-tests-02


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From nobody Sat May  7 21:56:49 2016
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From: "Xiaoqing Zhu (xiaoqzhu)" <xiaoqzhu@cisco.com>
To: "rmcat@ietf.org" <rmcat@ietf.org>
Thread-Topic: [rmcat] I-D Action: draft-ietf-rmcat-wireless-tests-02.txt
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Subject: Re: [rmcat] I-D Action: draft-ietf-rmcat-wireless-tests-02.txt
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Hi,=0A=
=0A=
This new version contains some updates on the Wi-Fi test cases (Section 4):=
 =0A=
* Revised PHY- and MAC- layer configurations in the test setup to reflect 8=
02.11n=0A=
* Added the two new test cases presented at IETF-95: varying number of RMCA=
T clients and RMCAT clients against dynamic background traffic (CBR over UD=
P). =0A=
=0A=
Thanks,=0A=
Xiaoqing=0A=
=0A=
________________________________________=0A=
From: rmcat <rmcat-bounces@ietf.org> on behalf of internet-drafts@ietf.org =
<internet-drafts@ietf.org>=0A=
Sent: Saturday, May 7, 2016 11:51 PM=0A=
To: i-d-announce@ietf.org=0A=
Cc: rmcat@ietf.org=0A=
Subject: [rmcat] I-D Action: draft-ietf-rmcat-wireless-tests-02.txt=0A=
=0A=
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.=0A=
This draft is a work item of the RTP Media Congestion Avoidance Techniques =
of the IETF.=0A=
=0A=
        Title           : Evaluation Test Cases for Interactive Real-Time M=
edia over Wireless Networks=0A=
        Authors         : Zaheduzzaman Sarker=0A=
                          Ingemar Johansson=0A=
                          Xiaoqing Zhu=0A=
                          Jiantao Fu=0A=
                          Wei-Tian Tan=0A=
                          Michael A. Ramalho=0A=
        Filename        : draft-ietf-rmcat-wireless-tests-02.txt=0A=
        Pages           : 21=0A=
        Date            : 2016-05-07=0A=
=0A=
Abstract:=0A=
   It is evident that to ensure seamless and robust user experience=0A=
   across all type of access networks multimedia communication suits=0A=
   should adapt to the changing network conditions.  There is an ongoing=0A=
   effort in IETF RMCAT working group to standardize rate adaptive=0A=
   algorithm(s) to be used in the real-time interactive communication.=0A=
   In this document test cases are described to evaluate the=0A=
   performances of the proposed endpoint adaptation solutions in LTE=0A=
   networks and Wi-Fi networks.  The proposed algorithms should be=0A=
   evaluated using the test cases defined in this document to select=0A=
   most optimal solutions.=0A=
=0A=
=0A=
The IETF datatracker status page for this draft is:=0A=
https://datatracker.ietf.org/doc/draft-ietf-rmcat-wireless-tests/=0A=
=0A=
There's also a htmlized version available at:=0A=
https://tools.ietf.org/html/draft-ietf-rmcat-wireless-tests-02=0A=
=0A=
A diff from the previous version is available at:=0A=
https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rmcat-wireless-tests-02=0A=
=0A=
=0A=
Please note that it may take a couple of minutes from the time of submissio=
n=0A=
until the htmlized version and diff are available at tools.ietf.org.=0A=
=0A=
Internet-Drafts are also available by anonymous FTP at:=0A=
ftp://ftp.ietf.org/internet-drafts/=0A=
=0A=


From nobody Thu May 19 11:33:24 2016
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From: Bob Briscoe <ietf@bobbriscoe.net>
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Date: Thu, 19 May 2016 19:29:37 +0100
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Cc: TCP Prague List <tcpPrague@ietf.org>
Subject: [rmcat] L4S/DualQ BoF-forming on tcpprague list: pls air your concerns
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Multiple IETF Transport mailing lists.

This is to announce that discussions are proceeding on the 
tcpprague@ietf.org list [see archive 
<https://www.ietf.org/mailman/listinfo/tcpprague>] about organising a 
BoF on Low Latency Low Loss Scalable throughput (L4S) service, for the 
IETF-96 Berlin timeframe.
The proposed L4S BoF will be about the full implications of L4S - on 
ECN, SCTP, RMCAT, AQM, etc. Despite the name of the mailing list, it is 
not just about so-called "TCP Prague" and its implications solely on TCP.

If you reply to this announcement, please cut all the mailing lists from 
the distr except tcpprague.

Anyone who is sceptical of the L4S approach, or its safety for the 
public Internet, please air your concerns. Ideally now on the tcpprague 
list, but certainly at or preferably before the proposed BoF in Berlin.
If you prefer to air your concerns on your favourite WG mailing list, 
pls at least ensure the tcpprague list is aware.

You may know L4S as the DualQ Coupled AQM. Background info on L4S 
including I-Ds, code, videos, etc. can be found here:
https://riteproject.eu/dctth/

Cheers




Bob

>
> -------- Forwarded Message --------
> Subject: 	Volunteers pls: L4S non-WG-forming BoF proposal cut-off 
> approaching
> Date: 	Wed, 18 May 2016 12:47:26 +0100
> From: 	Bob Briscoe <ietf@bobbriscoe.net>
> To: 	TCP Prague List <tcpPrague@ietf.org>
>
>
>
> Folks,
>
> At the Low Latency, Low Loss Scalable throughput (L4S) Bar Bof in 
> Buenos Aires, there was support for a BoF about L4S, In the IETF-96 
> (Berlin) time-frame.
> It was decided to initially use this ML, even tho the scope of L4S is 
> wider than TCP Prague.
> Consensus was to aim for a non-WG-forming BoF, to demonstrate 
> willingness to work on the pieces in existing IETF WGs, and to 
> co-ordinate approaches to each WG.
>
> The cut-off is now 2.5 weeks away.
> *2016-06-03 (Friday):* Cut-off date for BOF proposal requests to Area 
> Directors at UTC 23:59.
>
> To organise a successful BoF (referring to rfc5434 
> <https://tools.ietf.org/html/rfc5434>), we need volunteers for the 
> following tasks:
> #/ Draft a statement of problem & scope
> #/ List planned deliverables (incl. implementation, spec drafting), 
> milestones and target WG (also identify any critical interdependencies)
> #/ Identify and draft any necessary changes to WG charters, to cover 
> the above deliverables
> #/ Discuss with relevant WG chairs and ADs [see background below]
> #/ Identify volunteers (pref before the BoF) planning to work on each 
> deliverable
>
> Some people have already volunteered in general to help with arranging 
> the BoF, but we now need to get specific.
>
> Let's get volunteers for the above priority tasks first, but for 
> completeness I'll also list the formalities we have to do:
> #/ Decide on name of BoF (and mailing list)
> #/ BoF facilitators - chairs, scribes, etc
> #/ Draft the BoF agenda, incl. time constraints
> #/ Decide on the BoF questions
> #/ Estimate attendance, list conflicts, decide duration
> #/ Submit a formal request for the BoF via
> http://www.ietf.org/instructions/MTG-SLOTS.html and
> http://www.ietf.org/ietf/1bof-procedures.txt
>
> I'll start off co-ordinating and chasing all the above, but contact me 
> if you would like to take on the co-ordinating task.
>
> *Background work so far**
> *
> The idea of this BoF has been floated with WG chairs and ADs for some 
> time now (since Nov'15 in Yokohama), except AFAIK, we haven't talked 
> with the RMCAT chairs.
> Also we have already done considerable work on defining the problem 
> while writing various drafts and papers about L4S.
>
> But, that was just a small group of co-authors (me, Koen, Inton, 
> Olga). There's still a lot to do to build wider consensus.
> We can go ahead with a BoF scheduling request even if disagreements on 
> scope/problem remain.
> But before the BoF itself, we need to have any such disagreements 
> ironed out.
>
> At the Buenos Aires Bar Bof, we put up a todo list of pieces that will 
> need to be worked on:
> See http://www.bobbriscoe.net/presents/1604ietf/1604-l4s-bar-bof.pdf 
> particularly slides 9 & 10
> and the target WGs for each were:
> * TSVWG (the L4S identifier)
> * AQM (the AQM - the name gives the clue ;)
> * TCPM (the DCTCP-like congestion control, covering SCTP)
> * RMCAT (L4S variants of real-time congestion controls, or at least 
> standardise existing controls using the identifier)
>
> I believe, formally, each WG has to decide to charter additional work 
> on its own ML.
>
> [Since then, Ingemar has pointed out on this list that we also need to 
> get AQM & ECN-marking in radio networks, and that will require working 
> with/through other SDOs and/or developer groups that deal with the MAC 
> layer .
> V. important, but let's push that to a separate thread.]
>
> Many of the pieces have their own rationale, independent of L4S. The 
> idea of the BoF is to draw the bigger picture that motivates the work.
> That doesn't preclude incremental work on the pieces for those who are 
> not motivated by a big picture.
>
> Cheers
>
>
>
> Bob




-- 
________________________________________________________________
Bob Briscoe                               http://bobbriscoe.net/


--------------070606010507000205050305
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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=utf-8">
  </head>
  <body text="#000000" bgcolor="#FFFFFF">
    Multiple IETF Transport mailing lists.<br>
    <br>
    This is to announce that discussions are proceeding on the
    <a class="moz-txt-link-abbreviated" href="mailto:tcpprague@ietf.org">tcpprague@ietf.org</a> list [see <a
      href="https://www.ietf.org/mailman/listinfo/tcpprague">archive</a>]
    about organising a BoF on Low Latency Low Loss Scalable throughput
    (L4S) service, for the IETF-96 Berlin timeframe.<br>
    The proposed L4S BoF will be about the full implications of L4S - on
    ECN, SCTP, RMCAT, AQM, etc. Despite the name of the mailing list, it
    is not just about so-called "TCP Prague" and its implications solely
    on TCP.<br>
    <br>
    If you reply to this announcement, please cut all the mailing lists
    from the distr except tcpprague.<br>
    <br>
    Anyone who is sceptical of the L4S approach, or its safety for the
    public Internet, please air your concerns. Ideally now on the
    tcpprague list, but certainly at or preferably before the proposed
    BoF in Berlin. <br>
    If you prefer to air your concerns on your favourite WG mailing
    list, pls at least ensure the tcpprague list is aware.<br>
    <br>
    You may know L4S as the DualQ Coupled AQM. Background info on L4S
    including I-Ds, code, videos, etc. can be found here:<br>
    <a class="moz-txt-link-freetext" href="https://riteproject.eu/dctth/">https://riteproject.eu/dctth/</a><br>
    <br>
    Cheers<br>
    <br>
    <br>
    <br>
    <br>
    Bob<br>
    <br>
    <blockquote type="cite"><br>
      -------- Forwarded Message --------
      <table class="moz-email-headers-table" border="0" cellpadding="0"
        cellspacing="0">
        <tbody>
          <tr>
            <th nowrap="nowrap" valign="BASELINE" align="RIGHT">Subject:
            </th>
            <td>Volunteers pls: L4S non-WG-forming BoF proposal cut-off
              approaching</td>
          </tr>
          <tr>
            <th nowrap="nowrap" valign="BASELINE" align="RIGHT">Date: </th>
            <td>Wed, 18 May 2016 12:47:26 +0100</td>
          </tr>
          <tr>
            <th nowrap="nowrap" valign="BASELINE" align="RIGHT">From: </th>
            <td>Bob Briscoe <a class="moz-txt-link-rfc2396E" href="mailto:ietf@bobbriscoe.net">&lt;ietf@bobbriscoe.net&gt;</a></td>
          </tr>
          <tr>
            <th nowrap="nowrap" valign="BASELINE" align="RIGHT">To: </th>
            <td>TCP Prague List <a class="moz-txt-link-rfc2396E" href="mailto:tcpPrague@ietf.org">&lt;tcpPrague@ietf.org&gt;</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      Folks,<br>
      <br>
      At the Low Latency, Low Loss Scalable throughput (L4S) Bar Bof in
      Buenos Aires, there was support for a BoF about L4S, In the
      IETF-96 (Berlin) time-frame.<br>
      It was decided to initially use this ML, even tho the scope of L4S
      is wider than TCP Prague.<br>
      Consensus was to aim for a non-WG-forming BoF, to demonstrate
      willingness to work on the pieces in existing IETF WGs, and to
      co-ordinate approaches to each WG.<br>
      <br>
      The cut-off is now 2.5 weeks away. <br>
      *2016-06-03 (Friday):* Cut-off date for BOF proposal requests to
      Area Directors at UTC 23:59. <br>
      <br>
      To organise a successful BoF (referring to <a
        moz-do-not-send="true"
        href="https://tools.ietf.org/html/rfc5434">rfc5434</a>), we need
      volunteers for the following tasks:<br>
      #/ Draft a statement of problem &amp; scope<br>
      #/ List planned deliverables (incl. implementation, spec
      drafting), milestones and target WG (also identify any critical
      interdependencies)<br>
      #/ Identify and draft any necessary changes to WG charters, to
      cover the above deliverables<br>
      #/ Discuss with relevant WG chairs and ADs [see background below]<br>
      #/ Identify volunteers (pref before the BoF) planning to work on
      each deliverable<br>
      <br>
      Some people have already volunteered in general to help with
      arranging the BoF, but we now need to get specific.<br>
      <br>
      Let's get volunteers for the above priority tasks first, but for
      completeness I'll also list the formalities we have to do:<br>
      #/ Decide on name of BoF (and mailing list)<br>
      #/ BoF facilitators - chairs, scribes, etc<br>
      #/ Draft the BoF agenda, incl. time constraints<br>
      #/ Decide on the BoF questions<br>
      #/ Estimate attendance, list conflicts, decide duration<br>
      #/ Submit a formal request for the BoF via <br>
           <a moz-do-not-send="true" class="moz-txt-link-freetext"
        href="http://www.ietf.org/instructions/MTG-SLOTS.html">http://www.ietf.org/instructions/MTG-SLOTS.html</a>
      and<br>
           <a moz-do-not-send="true" class="moz-txt-link-freetext"
        href="http://www.ietf.org/ietf/1bof-procedures.txt">http://www.ietf.org/ietf/1bof-procedures.txt</a><br>
      <br>
      I'll start off co-ordinating and chasing all the above, but
      contact me if you would like to take on the co-ordinating task.<br>
      <br>
      <b>Background work so far</b><b><br>
      </b><br>
      The idea of this BoF has been floated with WG chairs and ADs for
      some time now (since Nov'15 in Yokohama), except AFAIK, we haven't
      talked with the RMCAT chairs.<br>
      Also we have already done considerable work on defining the
      problem while writing various drafts and papers about L4S.<br>
      <br>
      But, that was just a small group of co-authors (me, Koen, Inton,
      Olga). There's still a lot to do to build wider consensus.<br>
      We can go ahead with a BoF scheduling request even if
      disagreements on scope/problem remain. <br>
      But before the BoF itself, we need to have any such disagreements
      ironed out.<br>
      <br>
      At the Buenos Aires Bar Bof, we put up a todo list of pieces that
      will need to be worked on:<br>
      See <a moz-do-not-send="true" class="moz-txt-link-freetext"
        href="http://www.bobbriscoe.net/presents/1604ietf/1604-l4s-bar-bof.pdf">http://www.bobbriscoe.net/presents/1604ietf/1604-l4s-bar-bof.pdf</a>
      particularly slides 9 &amp; 10 <br>
      and the target WGs for each were:<br>
      * TSVWG (the L4S identifier)<br>
      * AQM (the AQM - the name gives the clue ;)<br>
      * TCPM (the DCTCP-like congestion control, covering SCTP)<br>
      * RMCAT (L4S variants of real-time congestion controls, or at
      least standardise existing controls using the identifier)<br>
      <br>
      I believe, formally, each WG has to decide to charter additional
      work on its own ML.<br>
      <br>
      [Since then, Ingemar has pointed out on this list that we also
      need to get AQM &amp; ECN-marking in radio networks, and that will
      require working with/through other SDOs and/or developer groups
      that deal with the MAC layer . <br>
      V. important, but let's push that to a separate thread.]<br>
      <br>
      Many of the pieces have their own rationale, independent of L4S.
      The idea of the BoF is to draw the bigger picture that motivates
      the work. <br>
      That doesn't preclude incremental work on the pieces for those who
      are not motivated by a big picture.<br>
      <br>
      Cheers<br>
      <br>
      <br>
      <br>
      Bob</blockquote>
    <br>
    <br>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
________________________________________________________________
Bob Briscoe                               <a class="moz-txt-link-freetext" href="http://bobbriscoe.net/">http://bobbriscoe.net/</a></pre>
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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
To: "rmcat WG (rmcat@ietf.org)" <rmcat@ietf.org>
Thread-Topic: SCReAM status and way forward
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Date: Fri, 20 May 2016 10:51:31 +0000
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Subject: [rmcat] SCReAM status and way forward
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Hi

=20

The SCReAM congestion control algorithm has undergone an update after a
review by Mirja K=FChlewind, Stefan Holmer and Xiaoqing Zhu (thanks!).=20

There will likely be an additional update submitted in a few weeks.=20

=20

I have a few questions :

Q1:

Is this draft is now ready for WGLC ?=20

=20

Q2:

Which simulation data, if any, is required to get the draft into WGLC ?. =
I
understand that it is now sufficient with simulation results for the =
test
cases in section 5 in
(https://tools.ietf.org/html/draft-ietf-rmcat-eval-test-03 ), the =
wireless
test cases are optional. Is this interpretation correct ?.

=20

Q3:

Running code for SCReAM has been published quite a while ago, both as a
standalone implementation and as a part of OpenWebRTC, I have however =
not
seen any reaction or questions or other signs that indicate that this is =
of
interest to this group. Has anybody looked at this code, experimented =
with
it, or integrated it in one form or the other ?. Does anybody plan to do
anything with the code ?

=20

/Ingemar

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D

Ingemar Johansson  M.Sc.=20

Master Researcher

=20

Ericsson AB

Wireless Access Networks

Labratoriegr=E4nd 11

971 28, Lule=E5, Sweden

Phone +46-1071 43042

SMS/MMS +46-73 078 3289

 <mailto:ingemar.s.johansson@ericsson.com> =
ingemar.s.johansson@ericsson.com

www.ericsson.com

=20

           This report, by it=B4s very length,=20

 defends itself against the risk of being read

                       Winston Churchill

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D

=20


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<html xmlns:v=3D"urn:schemas-microsoft-com:vml" =
xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
xmlns:w=3D"urn:schemas-microsoft-com:office:word" =
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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p =
class=3DMsoNormal>Hi<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>The SCReAM =
congestion control algorithm has undergone an update after a review by =
Mirja K=FChlewind, Stefan Holmer and Xiaoqing Zhu (thanks!). =
<o:p></o:p></p><p class=3DMsoNormal>There will likely be an additional =
update submitted in a few weeks. <o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>I have a few =
questions :<o:p></o:p></p><p class=3DMsoNormal>Q1:<o:p></o:p></p><p =
class=3DMsoNormal>Is this draft is now ready for WGLC ? =
<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>Q2:<o:p></o:p></p><p class=3DMsoNormal>Which =
simulation data, if any, is required to get the draft into WGLC ?. I =
understand that it is now sufficient with simulation results for the =
test cases in section 5 in (<a =
href=3D"https://tools.ietf.org/html/draft-ietf-rmcat-eval-test-03">https:=
//tools.ietf.org/html/draft-ietf-rmcat-eval-test-03</a> ), the wireless =
test cases are optional. Is this interpretation correct =
?.<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>Q3:<o:p></o:p></p><p class=3DMsoNormal>Running code =
for SCReAM has been published quite a while ago, both as a standalone =
implementation and as a part of OpenWebRTC, I have however not seen any =
reaction or questions or other signs that indicate that this is of =
interest to this group. Has anybody looked at this code, experimented =
with it, or integrated it in one form or the other ?. Does anybody plan =
to do anything with the code ?<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>/Ingemar<o:p></o:p></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Ingemar =
Johansson&nbsp; M.Sc. <o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Master =
Researcher<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p><p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Ericsson =
AB<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Wireless =
Access Networks<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Labratoriegr=E4=
nd 11<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>971 28, =
Lule=E5, Sweden<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Phone =
+46-1071 43042<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>SMS/MMS =
+46-73 078 3289<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DSV =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><a =
href=3D"mailto:ingemar.s.johansson@ericsson.com"><span lang=3DEN-US =
style=3D'color:blue'>ingemar.s.johansson@ericsson.com</span></a></span><s=
pan =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></=
span></p><p class=3DMsoNormal style=3D'text-autospace:none'><span =
lang=3DSV style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><a =
href=3D"www.ericsson.com"><span lang=3DEN-US =
style=3D'color:blue'>www.ericsson.com</span></a></span><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></=
span></p><p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p><p class=3DMsoNormal style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0 This report, by it=B4s very length, =
<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>=A0defends =
itself against the risk of being read<o:p></o:p></span></p><p =
class=3DMsoNormal> =
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0Wins=
ton Churchill<o:p></o:p></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span lang=3DSV =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D<o:p></o:p></span></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></body></html>
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To: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>, "stefan@webrtc.org" <stefan@webrtc.org>, "randell-ietf@jesup.org" <randell-ietf@jesup.org>
References: <81564C0D7D4D2A4B9A86C8C7404A13DA43DC0906@ESESSMB205.ericsson.se> <CAEdus3L9Zomqn5Mt7wTvMschP8UDz5_sbg=V4kVNsyhKJWdnKA@mail.gmail.com> <81564C0D7D4D2A4B9A86C8C7404A13DA43DC10BF@ESESSMB205.ericsson.se> <CAEdus3L1H=gjCZxAFf8HxYLaRHrY0DztrTUq_rr_S63VMXDobw@mail.gmail.com> <81564C0D7D4D2A4B9A86C8C7404A13DA43DC1402@ESESSMB205.ericsson.se>
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Date: Tue, 24 May 2016 17:35:08 +0200
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Subject: Re: [rmcat] Live slide show modeling
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--------------020703050208050303060002
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All,

We would like to announce to the list that we have implemented a new 
synthetic codec C++ class (SimpleContentSharingCodec), which follows the 
guidelines that Stefan and Ingemar described in this email thread.

You can find it in the latest version of our repo: 
https://github.com/cisco/syncodecs

Let me know if you have questions on how to use it.

Regards,

Sergio


On 07/04/16 14:43, Ingemar Johansson S wrote:
>
> Did a quick shot where I created a frame size trace according to the 
> guidelines below , the probability of a large frame is 0.05.
>
> The bottleneck BW is limited to 500kbps, RTT=100ms
>
> Please see attached png.
>
> I did not optimize SCReAM for this, there is a possibility that one 
> need to turn off the feature that clears the transmit queue if it 
> becomes too large, this feature can then be defined as optional on a 
> SCReAM-stream level.
>
> Another observation is that CWND is estimated too high between ~12 and 
> 40s. So there is probably room for improvement but at least it does 
> not break down.
>
> I also attached the frame size trace, realize that the 1^st frame 
> should probably be large.
>
> /Ingemar
>
> *From:*Stefan Holmer [mailto:stefan@webrtc.org]
> *Sent:* den 7 april 2016 12:59
> *To:* Ingemar Johansson S; randell-ietf@jesup.org
> *Cc:* rmcat@ietf.org
> *Subject:* Re: Live slide show modeling
>
> Right. My experience is that during the periods of static content, the 
> congestion controller may run into problems due to being rate limited 
> and network jitter. Once it's time to transmit a huge frame the 
> rate/cwnd is very low and it takes several seconds to send the frame 
> even though the connection is typically good.
>
> On Thu, Apr 7, 2016 at 11:54 AM Ingemar Johansson S 
> <ingemar.s.johansson@ericsson.com 
> <mailto:ingemar.s.johansson@ericsson.com>> wrote:
>
>     Hi
>
>     Ok, thanks, I assume that rate adaptation is not really applicable
>     here ?. Instead it is more a task to verify that the congestion
>     control does not transmit more than what fits in the bottleneck,
>     to avoid excess packet drops or delay ?
>
>     /Ingemar
>
>     *From:*Stefan Holmer [mailto:stefan@webrtc.org
>     <mailto:stefan@webrtc.org>]
>     *Sent:* den 7 april 2016 10:27
>     *To:* Ingemar Johansson S; Randell Jesup (randell-ietf@jesup.org
>     <mailto:randell-ietf@jesup.org>)
>     *Cc:* rmcat WG (rmcat@ietf.org <mailto:rmcat@ietf.org>)
>     *Subject:* Re: Live slide show modeling
>
>     I'd model it as a low framerate (~5 fps) stream, where with some
>     probability (maybe between 0.005 and 0.05) a frame is large due to
>     a slide change (say 50-400 packets). In between slide changes the
>     frames can be compressed to a single packet.
>
>     /Stefan
>
>     On Thu, Apr 7, 2016 at 8:39 AM Ingemar Johansson S
>     <ingemar.s.johansson@ericsson.com
>     <mailto:ingemar.s.johansson@ericsson.com>> wrote:
>
>         Hi
>
>         Do you have a reasonably rough idea of how to model the
>         transmission of a live slide show.
>
>         I can, from the discussion on the chat, imagine something like
>
>         Frame period : 1s, or 5s, something else ?
>
>         Frame size : ? a 1920x1080 jpg compressed image is something
>         like 500+ kB ?
>
>         /Ingemar
>
>         ==================================
>
>         Ingemar Johansson  M.Sc.
>
>         Master Researcher
>
>         Ericsson AB
>
>         Wireless Access Networks
>
>         Labratoriegränd 11
>
>         971 28, Luleå, Sweden
>
>         Phone +46-1071 43042
>
>         SMS/MMS +46-73 078 3289
>
>         ingemar.s.johansson@ericsson.com
>         <mailto:ingemar.s.johansson@ericsson.com>
>
>         www.ericsson.com <http://www.ericsson.com>
>
>         This report, by it´s very length,
>
>          defends itself against the risk of being read
>
>                                Winston Churchill
>
>         ==================================
>


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    <div class="moz-cite-prefix"><br>
      All,<br>
      <br>
      We would like to announce to the list that we have implemented a
      new synthetic codec C++ class (SimpleContentSharingCodec), which
      follows the guidelines that Stefan and Ingemar described in this
      email thread.<br>
      <br>
      You can find it in the latest version of our repo:
      <a class="moz-txt-link-freetext" href="https://github.com/cisco/syncodecs">https://github.com/cisco/syncodecs</a><br>
      <br>
      Let me know if you have questions on how to use it.<br>
      <br>
      Regards,<br>
      <br>
      Sergio<br>
      <br>
      <br>
      On 07/04/16 14:43, Ingemar Johansson S wrote:<br>
    </div>
    <blockquote
cite="mid:%3C81564C0D7D4D2A4B9A86C8C7404A13DA43DC1402@ESESSMB205.ericsson.se%3E"
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      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Did
            a quick shot where I created a frame size trace according to
            the guidelines below , the probability of a large frame is
            0.05.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">The
            bottleneck BW is limited to 500kbps, RTT=100ms<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Please
            see attached png. <o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I
            did not optimize SCReAM for this, there is a possibility
            that one need to turn off the feature that clears the
            transmit queue if it becomes too large, this feature can
            then be defined as optional on a SCReAM-stream level.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Another
            observation is that CWND is estimated too high between ~12
            and 40s. So there is probably room for improvement but at
            least it does not break down.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I
            also attached the frame size trace, realize that the 1<sup>st</sup>
            frame should probably be large.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">/Ingemar
            <o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p> </o:p></span></p>
        <div style="border:none;border-left:solid blue 1.5pt;padding:0cm
          0cm 0cm 4.0pt">
          <div>
            <div style="border:none;border-top:solid #B5C4DF
              1.0pt;padding:3.0pt 0cm 0cm 0cm">
              <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">
                  Stefan Holmer [<a class="moz-txt-link-freetext" href="mailto:stefan@webrtc.org">mailto:stefan@webrtc.org</a>] <br>
                  <b>Sent:</b> den 7 april 2016 12:59<br>
                  <b>To:</b> Ingemar Johansson S; <a class="moz-txt-link-abbreviated" href="mailto:randell-ietf@jesup.org">randell-ietf@jesup.org</a><br>
                  <b>Cc:</b> <a class="moz-txt-link-abbreviated" href="mailto:rmcat@ietf.org">rmcat@ietf.org</a><br>
                  <b>Subject:</b> Re: Live slide show modeling<o:p></o:p></span></p>
            </div>
          </div>
          <p class="MsoNormal"><o:p> </o:p></p>
          <div>
            <p class="MsoNormal">Right. My experience is that during the
              periods of static content, the congestion controller may
              run into problems due to being rate limited and network
              jitter. Once it's time to transmit a huge frame the
              rate/cwnd is very low and it takes several seconds to send
              the frame even though the connection is typically good.<o:p></o:p></p>
          </div>
          <p class="MsoNormal"><o:p> </o:p></p>
          <div>
            <div>
              <p class="MsoNormal">On Thu, Apr 7, 2016 at 11:54 AM
                Ingemar Johansson S &lt;<a moz-do-not-send="true"
                  href="mailto:ingemar.s.johansson@ericsson.com">ingemar.s.johansson@ericsson.com</a>&gt;
                wrote:<o:p></o:p></p>
            </div>
            <blockquote style="border:none;border-left:solid #CCCCCC
              1.0pt;padding:0cm 0cm 0cm
              6.0pt;margin-left:4.8pt;margin-right:0cm">
              <div>
                <div>
                  <p class="MsoNormal"
                    style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi</span><o:p></o:p></p>
                  <p class="MsoNormal"
                    style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"> </span><o:p></o:p></p>
                  <p class="MsoNormal"
                    style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Ok,
                      thanks, I assume that rate adaptation is not
                      really applicable here ?. Instead it is more a
                      task to verify that the congestion control does
                      not transmit more than what fits in the
                      bottleneck, to avoid excess packet drops or delay
                      ?</span><o:p></o:p></p>
                  <p class="MsoNormal"
                    style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"> </span><o:p></o:p></p>
                  <p class="MsoNormal"
                    style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">/Ingemar</span><o:p></o:p></p>
                  <p class="MsoNormal"
                    style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"> </span><o:p></o:p></p>
                  <div style="border:none;border-left:solid blue
                    1.5pt;padding:0cm 0cm 0cm 4.0pt">
                    <div>
                      <div style="border:none;border-top:solid #B5C4DF
                        1.0pt;padding:3.0pt 0cm 0cm 0cm">
                        <p class="MsoNormal"
                          style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">
                            Stefan Holmer [mailto:<a
                              moz-do-not-send="true"
                              href="mailto:stefan@webrtc.org"
                              target="_blank"><a class="moz-txt-link-abbreviated" href="mailto:stefan@webrtc.org">stefan@webrtc.org</a></a>] <br>
                            <b>Sent:</b> den 7 april 2016 10:27<br>
                            <b>To:</b> Ingemar Johansson S; Randell
                            Jesup (<a moz-do-not-send="true"
                              href="mailto:randell-ietf@jesup.org"
                              target="_blank">randell-ietf@jesup.org</a>)<br>
                            <b>Cc:</b> rmcat WG (<a
                              moz-do-not-send="true"
                              href="mailto:rmcat@ietf.org"
                              target="_blank"><a class="moz-txt-link-abbreviated" href="mailto:rmcat@ietf.org">rmcat@ietf.org</a></a>)<br>
                            <b>Subject:</b> Re: Live slide show modeling</span><o:p></o:p></p>
                      </div>
                    </div>
                  </div>
                </div>
              </div>
              <div>
                <div>
                  <div style="border:none;border-left:solid blue
                    1.5pt;padding:0cm 0cm 0cm 4.0pt">
                    <p class="MsoNormal"
                      style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
                    <div>
                      <p class="MsoNormal"
                        style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">I'd
                        model it as a low framerate (~5 fps) stream,
                        where with some probability (maybe between 0.005
                        and 0.05) a frame is large due to a slide change
                        (say 50-400 packets). In between slide changes
                        the frames can be compressed to a single packet.<o:p></o:p></p>
                      <div>
                        <p class="MsoNormal"
                          style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
                      </div>
                      <div>
                        <p class="MsoNormal"
                          style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">/Stefan<o:p></o:p></p>
                      </div>
                    </div>
                    <p class="MsoNormal"
                      style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
                    <div>
                      <div>
                        <p class="MsoNormal"
                          style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">On
                          Thu, Apr 7, 2016 at 8:39 AM Ingemar Johansson
                          S &lt;<a moz-do-not-send="true"
                            href="mailto:ingemar.s.johansson@ericsson.com"
                            target="_blank">ingemar.s.johansson@ericsson.com</a>&gt;
                          wrote:<o:p></o:p></p>
                      </div>
                      <blockquote style="border:none;border-left:solid
                        #CCCCCC 1.0pt;padding:0cm 0cm 0cm
6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt">
                        <div>
                          <div>
                            <p class="MsoNormal"
                              style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Hi<o:p></o:p></p>
                            <p class="MsoNormal"
                              style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Do
                              you have a reasonably rough idea of how to
                              model the transmission of a live slide
                              show.<o:p></o:p></p>
                            <p class="MsoNormal"
                              style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">I
                              can, from the discussion on the chat,
                              imagine something like<o:p></o:p></p>
                            <p class="MsoNormal"
                              style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
                            <p class="MsoNormal"
                              style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Frame
                              period : 1s, or 5s, something else ?<o:p></o:p></p>
                            <p class="MsoNormal"
                              style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Frame
                              size : ? a 1920x1080 jpg compressed image
                              is something like 500+ kB ?<o:p></o:p></p>
                            <p class="MsoNormal"
                              style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
                            <p class="MsoNormal"
                              style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">/Ingemar<o:p></o:p></p>
                            <p class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;text-autospace:none"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">==================================</span><o:p></o:p></p>
                            <p class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;text-autospace:none"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Ingemar
                                Johansson  M.Sc. </span><o:p></o:p></p>
                            <p class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;text-autospace:none"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Master
                                Researcher</span><o:p></o:p></p>
                            <p class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;text-autospace:none"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"> </span><o:p></o:p></p>
                            <p class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;text-autospace:none"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Ericsson
                                AB</span><o:p></o:p></p>
                            <p class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;text-autospace:none"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Wireless
                                Access Networks</span><o:p></o:p></p>
                            <p class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;text-autospace:none"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Labratoriegränd
                                11</span><o:p></o:p></p>
                            <p class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;text-autospace:none"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">971
                                28, Luleå, Sweden</span><o:p></o:p></p>
                            <p class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;text-autospace:none"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Phone
                                +46-1071 43042</span><o:p></o:p></p>
                            <p class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;text-autospace:none"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">SMS/MMS
                                +46-73 078 3289</span><o:p></o:p></p>
                            <p class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;text-autospace:none"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"
                                lang="SV"><a moz-do-not-send="true"
                                  href="mailto:ingemar.s.johansson@ericsson.com"
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style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">          
                                This report, by it´s very length, </span><o:p></o:p></p>
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style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;text-autospace:none"><span
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                                itself against the risk of being read</span><o:p></o:p></p>
                            <p class="MsoNormal"
                              style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">                       Winston
                              Churchill<o:p></o:p></p>
                            <p class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;text-autospace:none"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;"
                                lang="SV">==================================</span><o:p></o:p></p>
                            <p class="MsoNormal"
                              style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
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            </blockquote>
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Subject: [rmcat]  SCReAM status and way forward
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Ingemar, for what it's worth, I forked the SCReAM code on github and integr=
ated it into an RTP library.
I ran some preliminary tests streaming video over my home ADSL line, but of=
ten found that the algorithm
struggled to stream at higher rates than the minimum (in my case I used 50k=
bps as the minimum).=20
If I understand correctly, the algorithm is designed for use over mobile ne=
tworks and not ADSL though.=20
If you're interested, perhaps we could investigate the behaviour of the alg=
orithm on other network types, etc.
It could be that some fine-tuning, etc is needed to make it work in such ca=
ses. Of course it could also be,
that I was not using your code as intended...

R.

> Q3:
> Running code for SCReAM has been published quite a while ago, both as a
> standalone implementation and as a part of OpenWebRTC, I have however not
> seen any reaction or questions or other signs that indicate that this is =
of
> interest to this group. Has anybody looked at this code, experimented with
> it, or integrated it in one form or the other ?. Does anybody plan to do
> anything with the code ?



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This message is subject to the CSIR's copyright terms and conditions, e-mai=
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r.html.

This message has been scanned for viruses and dangerous content by MailScan=
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and is believed to be clean.

Please consider the environment before printing this email.


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From: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>
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Hi Ralf

 

And thanks for the effort, worth a lot. 

I assume that you have used the C++ code
(https://github.com/EricssonResearch/scream) in your implementation?

 

50kbps is a very low bitrate for video, there are a few things that can
cause extra trouble. 

+ RTCP overhead. At bitrates this low it is sufficient with feedback
intervals around 200ms, furthermore reduced size RTCP (RFC5506) is highly
recommended, this will give you an RTCP overhead of ~3-5kbps, so a
recommendation is that you have a look at how much bandwidth RTCP consumes.
Is reduced size RTCP implemented in your RTP library ?, it is implemented in
the GStreamer libraries if you wish to have a look at how it is done.

 

+ RTP overhead. The function getTargetBitrate has a comment in the header
file

    /*

    * Get the target bitrate for stream with SSRC

    * NOTE!, Because SCReAM operates on RTP packets, the target bitrate will

    *  also include the RTP overhead. This means that a subsequent call to
set the 

    *  media coder target bitrate must subtract an estimate of the RTP +
framing 

    *  overhead. This is not critical for Video bitrates but can be
important 

    *  when SCReAM is used to congestion control e.g low bitrate audio
streams

    */

   This means that, if the size argument in the addTransmitted function
includes also the size of the RTP header (+ possible extensions) then the
computed bitrate is the bitrate with the RTP overhead included. This means
that you need to subtract the bitrate given by the RTP overhead when you
compute the video coder bitrate. The function to compute the video coder
bitrate is thus.

videoBitrate = getTargetBitrate(..)+rtpOverheadBitrate.

  A somewhat better alternative, used in the OpenWebRTC implementation is to
use the RTP payload size as input to the addTransmitted(..) function.

 

+ Video coder output. My grim experience is that the video coder does not
always output the desired target bitrate, variations around the target
bitrate is natural but in some cases I have seen that the output bitrate is
consistently 10-20% above the target bitrate, for that reason it could be
good to compute and log the video coder output bitrate. In some cases the
video coders are in a non-rate adaptive mode or have the internal rate
control tuned too slow, this is a tuning matter. I have fiddled a bit with
OpenH264 and VP8, which video codec do you use ?

 

I have only tried out these low bitrates with the experimental code, the
code is adapted for audio streaming so it should not be any problems to
congestion control video  over 50kbps bottlenecks. 

With the OpenWebRTC implementation I managed to push down the bottleneck
bitrate down to 200kbps. The problem there is that I did not have full
control over the RTP stack and that RTCP tended to consume too much
bandwidth.

 

That said, there are probably some yet to discover issues, it is not too
unlikely that you have found an issue in  SCReAM but one probably need to
rule out the  possible issues above before. 

 

Is your fork available somewhere ?, could be interesting to try it out as
you have integrated it into an RTP library.

 

Regards

Ingemar

 

 

 

"Ralf Globisch" <rglobisch@csir.co.za> Tue, 31 May 2016 19:14 UTC
<https://mailarchive.ietf.org/arch/search/?email_list=rmcat> Show header

Ingemar, for what it's worth, I forked the SCReAM code on github and
integrated it into an RTP library.
I ran some preliminary tests streaming video over my home ADSL line, but
often found that the algorithm
struggled to stream at higher rates than the minimum (in my case I used
50kbps as the minimum). 
If I understand correctly, the algorithm is designed for use over mobile
networks and not ADSL though. 
If you're interested, perhaps we could investigate the behaviour of the
algorithm on other network types, etc.
It could be that some fine-tuning, etc is needed to make it work in such
cases. Of course it could also be,
that I was not using your code as intended...
 

 

 


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vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>Hi Ralf<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>And thanks for the =
effort, worth a lot. <o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>I assume that you have used the C++ code =
(https://github.com/EricssonResearch/scream) in your =
implementation?<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>50kbps is a very low =
bitrate for video, there are a few things that can cause extra trouble. =
<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>+ RTCP overhead. At bitrates this low it is =
sufficient with feedback intervals around 200ms, furthermore reduced =
size RTCP (RFC5506) is highly recommended, this will give you an RTCP =
overhead of ~3-5kbps, so a recommendation is that you have a look at how =
much bandwidth RTCP consumes. Is reduced size RTCP implemented in your =
RTP library ?, it is implemented in the GStreamer libraries if you wish =
to have a look at how it is done.<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>+ RTP overhead. The =
function </span><span =
style=3D'font-size:9.5pt;font-family:Consolas;color:black;background:whit=
e;mso-highlight:white'>getTargetBitrate</span><span =
style=3D'color:#1F497D'> has a comment in the header =
file<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
style=3D'font-size:9.5pt;font-family:Consolas;color:black;background:whit=
e;mso-highlight:white'>&nbsp;&nbsp;&nbsp; </span><span =
style=3D'font-size:9.5pt;font-family:Consolas;color:green;background:whit=
e;mso-highlight:white'>/*</span><span =
style=3D'font-size:9.5pt;font-family:Consolas;color:black;background:whit=
e;mso-highlight:white'><o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
style=3D'font-size:9.5pt;font-family:Consolas;color:green;background:whit=
e;mso-highlight:white'>&nbsp;&nbsp;&nbsp; * Get the target bitrate for =
stream with SSRC</span><span =
style=3D'font-size:9.5pt;font-family:Consolas;color:black;background:whit=
e;mso-highlight:white'><o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
style=3D'font-size:9.5pt;font-family:Consolas;color:green;background:whit=
e;mso-highlight:white'>&nbsp;&nbsp;&nbsp; * NOTE!, Because SCReAM =
operates on RTP packets, the target bitrate will</span><span =
style=3D'font-size:9.5pt;font-family:Consolas;color:black;background:whit=
e;mso-highlight:white'><o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
style=3D'font-size:9.5pt;font-family:Consolas;color:green;background:whit=
e;mso-highlight:white'>&nbsp;&nbsp;&nbsp; *&nbsp; also include the RTP =
overhead. This means that a subsequent call to set the </span><span =
style=3D'font-size:9.5pt;font-family:Consolas;color:black;background:whit=
e;mso-highlight:white'><o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
style=3D'font-size:9.5pt;font-family:Consolas;color:green;background:whit=
e;mso-highlight:white'>&nbsp;&nbsp;&nbsp;&nbsp;*&nbsp; media coder =
target bitrate must subtract an estimate of the RTP + framing =
</span><span =
style=3D'font-size:9.5pt;font-family:Consolas;color:black;background:whit=
e;mso-highlight:white'><o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
style=3D'font-size:9.5pt;font-family:Consolas;color:green;background:whit=
e;mso-highlight:white'>&nbsp;&nbsp;&nbsp;&nbsp;*&nbsp; overhead. This is =
not critical for Video bitrates but can be important </span><span =
style=3D'font-size:9.5pt;font-family:Consolas;color:black;background:whit=
e;mso-highlight:white'><o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'text-autospace:none'><span =
style=3D'font-size:9.5pt;font-family:Consolas;color:green;background:whit=
e;mso-highlight:white'>&nbsp;&nbsp;&nbsp;&nbsp;*&nbsp; when SCReAM is =
used to congestion control e.g low bitrate audio streams</span><span =
style=3D'font-size:9.5pt;font-family:Consolas;color:black;background:whit=
e;mso-highlight:white'><o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:9.5pt;font-family:Consolas;color:green;background:whit=
e;mso-highlight:white'>&nbsp;&nbsp;&nbsp; */</span><span =
style=3D'color:#1F497D'><o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>&nbsp;&nbsp; This means that, if the size =
argument in the </span>addTransmitted <span =
style=3D'color:#1F497D'>function includes also the size of the RTP =
header (+ possible extensions) then the computed bitrate is the bitrate =
with the RTP overhead included. This means that you need to subtract the =
bitrate given by the RTP overhead when you compute the video coder =
bitrate. The function to compute the video coder bitrate is =
thus.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>videoBitrate =3D =
getTargetBitrate(..)+rtpOverheadBitrate.<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>&nbsp; A somewhat better =
alternative, used in the OpenWebRTC implementation is to use the RTP =
payload size as input to the addTransmitted(..) =
function.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>+ Video coder output. My =
grim experience is that the video coder does not always output the =
desired target bitrate, variations around the target bitrate is natural =
but in some cases I have seen that the output bitrate is consistently =
10-20% above the target bitrate, for that reason it could be good to =
compute and log the video coder output bitrate. In some cases the video =
coders are in a non-rate adaptive mode or have the internal rate control =
tuned too slow, this is a tuning matter. I have fiddled a bit with =
OpenH264 and VP8, which video codec do you use ?<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>I have only tried out =
these low bitrates with the experimental code, the code is adapted for =
audio streaming so it should not be any problems to congestion control =
video &nbsp;over 50kbps bottlenecks. <o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>With the OpenWebRTC =
implementation I managed to push down the bottleneck bitrate down to =
200kbps. The problem there is that I did not have full control over the =
RTP stack and that RTCP tended to consume too much =
bandwidth.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>That said, there are =
probably some yet to discover issues, it is not too unlikely that you =
have found an issue in &nbsp;SCReAM but one probably need to rule out =
the &nbsp;possible issues above before. <o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>Is your fork available =
somewhere ?, could be interesting to try it out as you have integrated =
it into an RTP library.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'>Regards<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'>Ingemar<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><div =
style=3D'border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt'><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3Ddarkgray =
style=3D'margin:0cm;margin-bottom:.0001pt;line-height:13.5pt;background:w=
hite;vertical-align:baseline'><span class=3Dpipe><span =
style=3D'font-size:9.0pt;font-family:"inherit","serif";color:#666666;bord=
er:none windowtext 1.0pt;padding:0cm'>&quot;Ralf Globisch&quot; &lt;<a =
href=3D"mailto:rglobisch@csir.co.za">rglobisch@csir.co.za</a>&gt;</span><=
/span><span class=3Dapple-converted-space><span =
style=3D'font-size:9.0pt;font-family:"Tahoma","sans-serif";color:#666666'=
>&nbsp;</span></span><span class=3Dpipe><span =
style=3D'font-size:9.0pt;font-family:"inherit","serif";color:#666666;bord=
er:none windowtext 1.0pt;padding:0cm'>Tue, 31 May 2016 19:14 =
UTC</span></span><span =
style=3D'font-size:9.0pt;font-family:"Tahoma","sans-serif";color:#666666'=
><a =
href=3D"https://mailarchive.ietf.org/arch/search/?email_list=3Drmcat" =
id=3Dtoggle><span =
style=3D'font-family:"inherit","serif";color:#5B80B2;border:none =
windowtext 1.0pt;padding:0cm'>Show =
header</span></a><o:p></o:p></span></p><pre =
style=3D'line-height:13.5pt;background:white;vertical-align:baseline;font=
-style:inherit;font-variant:inherit;font-weight:inherit;font-stretch: =
inherit;font-size:inherit;line-height:inherit;white-space:pre-wrap;word-w=
rap: break-word'><span =
style=3D'font-family:Courier;color:black'>Ingemar, for what it's worth, =
I forked the SCReAM code on github and integrated it into an RTP =
library.<o:p></o:p></span></pre><pre =
style=3D'line-height:13.5pt;background:white;vertical-align:baseline'><sp=
an style=3D'font-family:Courier;color:black'>I ran some preliminary =
tests streaming video over my home ADSL line, but often found that the =
algorithm<o:p></o:p></span></pre><pre =
style=3D'line-height:13.5pt;background:white;vertical-align:baseline'><sp=
an style=3D'font-family:Courier;color:black'>struggled to stream at =
higher rates than the minimum (in my case I used 50kbps as the minimum). =
<o:p></o:p></span></pre><pre =
style=3D'line-height:13.5pt;background:white;vertical-align:baseline'><sp=
an style=3D'font-family:Courier;color:black'>If I understand correctly, =
the algorithm is designed for use over mobile networks and not ADSL =
though. <o:p></o:p></span></pre><pre =
style=3D'line-height:13.5pt;background:white;vertical-align:baseline'><sp=
an style=3D'font-family:Courier;color:black'>If you're interested, =
perhaps we could investigate the behaviour of the algorithm on other =
network types, etc.<o:p></o:p></span></pre><pre =
style=3D'line-height:13.5pt;background:white;vertical-align:baseline'><sp=
an style=3D'font-family:Courier;color:black'>It could be that some =
fine-tuning, etc is needed to make it work in such cases. Of course it =
could also be,<o:p></o:p></span></pre><pre =
style=3D'line-height:13.5pt;background:white;vertical-align:baseline'><sp=
an style=3D'font-family:Courier;color:black'>that I was not using your =
code as intended...<o:p></o:p></span></pre><pre =
style=3D'line-height:13.5pt;background:white;vertical-align:baseline'><sp=
an =
style=3D'font-family:Courier;color:black'><o:p>&nbsp;</o:p></span></pre><=
p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></body></html>
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