Internet DRAFT - draft-aboba-rtpscale


     INTERNET-DRAFT                                           Bernard Aboba
     <draft-aboba-rtpscale-02.txt>                    Microsoft Corporation

                  Alternatives for Enhancing RTCP Scalability

     1.  Status of this Memo

     This document is an Internet-Draft.  Internet-Drafts are working docu-
     ments of the Internet Engineering Task Force (IETF),  its  areas,  and
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     The  distribution  of  this memo is unlimited.  It is filed as <draft-
     aboba-rtpscale-02.txt>, and  expires August 1, 1997. Please send  com-
     ments to the authors.

     2.  Abstract

     This  document  discusses current issues with RTCP scalability as well
     as describing the advantages and disadvantages of possible  solutions.

     3.  Requirements

     In  evaluating a solution to the RTCP scaling problem, it is important
     to have an understanding of the requirements that a proposed  solution
     must meet.  This document proposes four requirements:

          Decrease in convergence time
          Decrease in forwarding table entries
          Improvement in receiver reporting information
          Ability to adjust RTCP bandwidth fraction

     3.1.  Decrease in convergence time

     Currently RTCP relies on multicasting of receiver reports to establish
     a receiver-population estimate.  This shared-state  is  then  used  by
     receivers  to  establish  the  receiver  reporting  interval. The RTCP
     reporting algorithm described in [1], provides  for  an  initial  RTCP
     reporting  interval  randomized  on  [1.25,3.75].  While  the use of a

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     minimum waiting time for the initial report allows a receiver to  lis-
     ten  for  other  reports  prior  to  sending,  the reporting algorithm
     described in [1] does not mandate this. As a result,  reporting  hosts
     not  implementing  reconsideration (in which the reporting interval is
     readjusted prior to sending receiver reports) will send their  reports
     within  the  initially  calculated  interval, resulting in a potential
     flood of initial reports.

     3.2.  Decrease in forwarding table entries

     As defined in [1], RTCP messages are sent on the  same  group  as  the
     original  RTP transmission, but on adjacent ports. This is problematic
     for multicast routing protocols such as sparse mode PIM, which  deter-
     mine whether a given group will be hosted on the shared tree or source
     tree based on the sending rate. If an  RTP  group  is  placed  on  the
     source  tree,  then  if RTCP messages are sent to the same group, then
     each host sending RTCP receiver reports will result  in  a  forwarding
     table entry.

     However,  it should be noted that even if a shared-tree routing proto-
     col (such as CBT) is used, a problem will  still  exist  when  such  a
     domain  is  connected to a source-tree domain using a routing protocol
     such as MOSPF or DVMRP. This is because today there  is  no  mechanism
     for a multicast area border router to summarize RTCP activity within a
     multicast routing domain. Therefore if a shared tree  domain  is  con-
     nected  to  another  domain (such as a DVMRP domain) then packets from
     each RTCP source within the shared tree domain will need  to  be  for-
     warded into the source tree domain.

     This  problem  is  very common given that DVMRP functions today as the
     MBONE's multicast routing protocol. DVMRP generates  forwarding  cache
     entries  on  demand for each (source network, destination group) pair,
     and supports source-specific prune  messages. The  typical  forwarding
     cache expiration time is 200 seconds. Since for a large conference the
     RTCP reporting interval will typically  exceed  the  forwarding  cache
     expiration time shortly after conference initiation, only a portion of
     all (source network, destination group) pairs would  be  cached  by  a
     DVMRP  implementation  at  a given time. Nevertheless, the router will
     expend considerable resources in adding and flushing forwarding  cache
     entries,  as  well as in responding to source-specific prune requests.
     While future versions of DVMRP may prove more scalable, it is unlikely
     that these versions will be widely deployed within the next 18 months.
     As a result, summarization messages are desirable in multicast routing
     protocols,  as  well  as  a decrease in the number senders on the RTCP

     3.3.  Improvement in receiver reporting information

     The RTCP receiver report mechanism provides important  information  on
     listenership,  reception  quality,  and  bandwidth  availability. This
     information is important useful both for engineering and business pur-
     poses.   Engineers  use  reception  quality  information  to  diagnose

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     problems with the network. Sending systems can use  reception  quality
     and  bandwidth availability information to modify transmission parame-
     ters such as compression ratio and sending rate. Business  people  use
     information  on  listenership  to  track the size of the audience, and
     reception quality information to measure the quality of the user expe-

     Since  in  large  conferences  the algorithm described in [1] leads to
     infrequent receiver reporting, it can be argued that it does not  meet
     the  requirements for timely receiver reporting. Therefore any scaling
     "improvement" should not hamper the ability to collect  this  informa-
     tion, and probably should be expected to result in improved reporting.

     3.4.  Ability to adjust RTCP bandwidth fraction

     In low or asymmetric bandwidth situations,  the  fraction  of  session
     bandwidth  reserved  for  RTCP may need to be decreased below the five
     percent specified in [1]. For example, in Cable Internet it is  common
     for  downstream  bandwidth  to  be  a  factor  of  twenty greater than
     upstream bandwidth; allowing five percent of downstream  bandwidth  to
     be  used for receiver reporting upstream would result in congestion of
     the upstream channel. In such situations it is necessary to allow  the
     bandwidth fraction to be adjusted.

     4.  Scaling alternatives

     Several  alternatives have been proposed for improving the scalability
     of RTCP. These include:

          Turning off RTCP receiver reports
          Improved convergence algorithms
          Use of separate multicast groups for RTP and RTCP
          Use of separate multicast groups for receiver and sender reports
          Unicasting of receiver reports
          Selective reporting
          Use of RTCP BYE messages
          Use of TTL scoping and summary messages
          Use of RTP translators

     These alternatives are discussed in turn.

     4.1.  Turning off RTCP receiver reports

     In some applications (such as satellite transmission) there may be  no
     upstream  channel  for  transmission  of  RTCP  receiver reports. As a
     result, it may be desirable to allow receiver reporting to  be  turned
     off.  Since  there  is no receiver reporting, there would be no way to
     estimate the receiver population.

     Influence on RTCP receivers could be exercised via the  SDP  announce-
     ment  mechanism.  For  example,  Mark  Handley  has  proposed that the

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     session profile specified in SDP via the "m=" line be used  to  define
     the  desired  RTCP  behavior. This would allow for turning off of RTCP
     receiver reports, or other modifications to reporting behavior.

     While this proposal would eliminate problems relating  to  RTCP  band-
     width  usage in dialup and asymmetric systems, it would deprive inter-
     ested parties of information on listenership  and  reception  quality.
     This  will  prevent senders from making adjustments to their transmis-
     sion parameters, and would also prevent RTP  monitors  from  providing
     listenership estimates needed to justify advertising rates.

     However,  it  is  not clear that these arguments are persuasive in all
     cases. RTCP receiver reports serve multiple purposes. One of these  is
     to provide information on listenership; another is to provide informa-
     tion on reception quality and bandwidth availability.  In  some  Cable
     Internet  systems,  it is possible to gain information on listenership
     via IGMP join and leave messages, since if the upstream and downstream
     channels  are separated receivers do not necessarily hear each other's
     join or leave messages. Furthermore, it can be argued that  the  state
     of  multicast  diagnostics  will eventually reach the point where RTCP
     receiver reports will no longer needed for diagnostic purposes.  Where
     receiver-based  rate  adjustment  is used, there may not be a need for
     the sender to adjust their transmission parameters.

     4.2.  Improved convergence algorithms

     Just as non-linear equation solvers can benefit  from  improved  algo-
     rithms,  it  may  be possible to improve RTP receiver population esti-
     mates  via  improvements  in  the  population  estimation   algorithm.
     Improvements  include  reconsideration, as well an initial  population
     estimate, and the incorporation of  additional  information  into  the

     The idea behind reconsideration, first proposed by Van Jacobson, is to
     improve the convergence time by calculating a new  estimate  of  group
     size  and  reporting  interval  prior to sending the RTCP report.  The
     report is only sent if the updated estimate is less than or  equal  to
     the  original  estimate;  otherwise  the  report  is  not sent and the
     reporting interval is adjusted to the new estimate.  Use of  reconsid-
     eration  greatly  reduces  the number of initial reports, since a host
     joining a conference already in progress will be brought  to  equilib-
     rium prior to sending its first report. As a result, implementation of
     reconsideration should be made mandatory in future versions of the RTP

     In  certain  circumstances, it may desirable to increase the dampening
     effect of reconsideration by increasing the initial minimum  reporting
     interval.  This will decrease the number of initial reports from sites
     with long delays, i.e. sites with a satellite  downlink  and  POTS  or
     ISDN  uplink,  as  well  as  to decrease the effect of sites reporting
     early merely due to statistical considerations.

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     Currently the receiver population estimate has an initial value of  1,
     but  it  is possible for an initial population estimate to be supplied
     in the session announcement.  Successive  population  estimates  could
     then  be  derived  via  an  averaging  of the initial estimate and the
     receiver's own estimate, so that the effect of  the  initial  estimate
     would  die  out  over time. However, as in nonlinear equation solving,
     use of an initial estimate is only helpful if it accurate and the ini-
     tial  estimate  is  likely  to prove less important than the algorithm
     which drives the system to steady state.  While  an  improved  initial
     population  estimate  would result in improved convergence times, were
     the initial estimate to be way off, convergence might be hindered.

     The rate of convergence may be improved by allowing  the  receiver  to
     use  information on the rate at which receiver reports are being sent.
     For example, due to bandwidth limitations, receivers on  higher  band-
     width connections have greater knowledge of the overall receiver popu-
     lation. Thus if a receiver were to note a receiver report from a  sys-
     tem  with  high  bandwidth  availability, that report could be weighed
     more heavily in determining the overall population estimate.

     It is also possible to incorporate the overall receiver reporting rate
     into  the  reporting interval calculation.  For example, if my current
     population estimate results in a receiver reporting interval of 3 min-
     utes, and I am receiving 200 receiver reports/minute, it may be desir-
     able to incorporate the rate of receiver reports into my calculations,
     increasing the reporting interval accordingly.

     However,  the  utility of taking rate information into account is lim-
     ited in the case of dialup clients, since they will only  be  able  to
     receive  a modest number of receiver reports/minute, and thus the rate
     at which receiver reports are flowing in may not be reflective of  the
     overall receiver population.

     While  improved  convergence  algorithms can result in marked improve-
     ments in convergence times,  and do result in more accurate population
     estimates,  they  do  not result in a lower number of forwarding table
     entries or senders to the RTCP receiver report group. They also do not
     influence the steady state bandwidth usage for RTCP reporting.

     4.3.  Use of separate multicast groups for RTP and RTCP

     Use  of separate multicast groups for RTP and RTCP results in improved
     scalability for multicasting routing protocols  such  as  sparse  mode
     PIM,  since the RTCP group can be placed on the shared tree due to its
     low sending rate. While this would markedly  decrease  the  number  of
     forwarding  cache entries, the router will nevertheless expend consid-
     erable resources in adding and flushing forwarding cache entries,  and
     setting  up  the individual hosts on the shared tree. For this reason,
     we do not believe that moving RTCP to an adjacent group will be suffi-
     cient  for  very  large  conferences. Note also that using an adjacent
     group for RTCP does not improve the scalability for DVMRP,  the  MBONE
     multicast   routing   protocol, since this protocol has no notion of a
     shared tree.

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     Use of separate multicast groups for RTP and RTCP does not affect con-
     vergence  times  or affect reporting accuracy. It also does not influ-
     ence the steady state bandwidth usage for RTCP reporting.

     4.4.  Use of separate multicast groups for receiver and sender reports

     Currently RTCP sender and receiver reports are sent to the same multi-
     cast group, and both RTP senders and receivers join this  group.  Were
     sender  and  receiver reports to be multicast on different groups, RTP
     receivers could be allowed to only join the sender report group,  thus
     allowing  them  to  avoid  listening  to  receiver report traffic. RTP
     senders would still join both groups in order to receive  feedback  on
     listenership  and  transmission  quality,  and  would  need to provide
     information on the estimated receiver  population  within  the  sender

     While  this  proposal would lessen the problem for receivers, it would
     not improve things for senders, and might even make them worse.  Since
     receivers  would  not  longer  be able to implement reconsideration as
     effectively, it would be likely that the senders would be deluged with
     initial  receiver  reports.  Use  of  separate groups for receiver and
     sender reports  would not result in a lower number of senders  on  the
     RTCP  receiver  report group, although it would decrease the number of
     forwarding table entries in protocol such as  sparse  mode  PIM.  This
     proposal  would eliminate the bandwidth used by receivers for incoming
     RTCP reports, but would not address the problem  with  upstream  band-
     width usage in asymmetric networks.

     4.5.  Unicasting of receiver reports

     Instead  of  multicasting  receiver reports, it is possible to unicast
     them back to the senders. This would  allow  RTP  listeners  to  avoid
     receiver  report  traffic, while RTP senders would still receive feed-
     back on listenership and transmission quality. In order to control the
     receiver  report  transmission  rate,  senders  would  need to provide
     information  on  the  estimated  receiver  population  within   sender

     While  this  proposal  would lessen the problem for receivers, as with
     the previous proposal, it might make things worse for  senders,  since
     receivers  would  no longer be able to reconsider as effectively. How-
     ever, it would eliminate multicasting of RTCP receiver reports,  which
     will be of benefit to overtaxed routers. This proposal would eliminate
     the bandwidth used by receivers for incoming RTCP reports,  but  would
     not  address  the  problem with upstream bandwidth usage in asymmetric

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     4.6.  Selective reporting

     In selective  reporting,  RTCP  receiver  reports  are  only  sent  by
     selected  hosts.  This  method results in a reduction in the number of
     RTCP senders, with attendant reduction in the  forwarding  tables.  It
     also is likely to result in lower convergence times. Assuming that the
     statistical sampling was adequate,  the  accuracy  and  timeliness  of
     reporting  need  not  be adversely affected. However, this method does
     not affect the steady state bandwidth allocated to RTCP.

     The selection can occur via a polling or random  selection  mechanism,
     or it can occur by self-selection. Generally, random selection is pre-
     ferred since self-selection brings with it the possibility of  report-
     ing  implosions.  For example, if receiver reports were only generated
     when packet loss exceeded  a  given  threshold,  then  congestion  and
     attendant  packet loss would result in a large number of reports, mak-
     ing the situation worse.

     Polling or random selection  methods,  while  they  show  considerable
     promise, need to address security concerns. For example, if it is pos-
     sible to get receivers to respond via a  polling  message,  then  that
     message should be authenticated to prevent leveraged denial of service

     4.7.  Use of RTCP BYE messages

     Given that receiver reporting intervals will tend to be very long  for
     large conferences, it can be argued that absent an emergency, it makes
     sense to provide information on listenership,  reception  quality  and
     bandwidth  availability  within  the RTCP Bye message. Thus on leaving
     the conference, the receiver would send a message  providing  informa-
     tion on the time period in which they joined the conference, the over-
     all reception quality and  other  information.  Conventional  receiver
     reports  would then either not be sent at all, or would be sent with a
     TTL of 1. While such an approach would lessen the  congestion  problem
     at  the beginning of a session, it would increase the size of the RTCP
     BYE message, resulting in a spike in RTCP bandwidth consumption at the
     end  of a session. Given that no receiver population estimate would be
     available, it appears that this approach could actually worsen conver-
     gence  times,  unless it were combined with some kind of summarization
     mechanism.  It would also not reduce the number of  RTCP  senders,  or
     reduce the steady state RTCP bandwidth fraction.

     4.8.  Use of TTL scoping and summary messages

     In this approach, RTCP receiver reports would be sent with a TTL of 1,
     and a designated  summarizer  would  be  elected  to  provide  summary
     reports with a larger TTL. This approach has the advantage of increas-
     ing the leverage of those RTCP receiver reports that are sent, and  is
     likely  to  be particularly effective for conferences in which member-
     ship is densely distributed. However, in sparsely distributed  confer-
     ences,  the  effect  of  summarization  will  be small unless multiple

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     levels of summarization are used. The designated summarizer would  not
     necesarily  be  a  router; it could also be another receiver, although
     this brings up the possibility  of  how  a  new  summarizer  could  be
     elected if the current summarizer leaves the conference.

     Since in this scheme receiver reports are not forwarded, non-summariz-
     ing RTP receivers should use  the  population  estimate  gleaned  from
     local scope reports to calculate their reporting interval. Summarizers
     and RTP senders will then use global  estimates  gleaned  from  global
     scope  summary reports to calculate their reporting interval. A recom-
     mended bandwidth allocation for reporting is  25  percent  for  sender
     reports,  25  percent for summary reports, and 50 percent for receiver

     Since this proposal decreases the  scope  of  RTCP  announcements,  it
     would  substantially  reduce the number of RTCP senders visible to the
     multicast backbone, and would decrease convergence times, since  those
     messages that were sent would include more information on the receiver
     population. This proposal would also address the problems of intercon-
     necting  multicast  routing  domains,  since the multicast area border
     router would be able to summarize RTCP  behavior  within  its  domain,
     rather  than  passing  along  all RTCP reports. However, it would also
     require substantial modifications in RTP client behavior.

     4.8.1.  Summary reports

     Via the use of summary reports, privacy can be provided while simulta-
     neously  providing  senders  with  improved  listenership and receiver
     quality reporting. This is possible because in the existing  implemen-
     tation  much of the information gained from receiver reports is redun-
     dant. For example, if congestion results in packets being dropped on a
     particular  link,  then  all  receivers downstream from that link will
     report the  problem. Redundant reporting obscures the  information  of
     greatest  interest,  which  is  information on global listenership and
     reception quality. Via introduction of summary reports, it is possible
     to  provide  more  accurate  and  timely reporting on listenership and
     reception quality, as well as to address issues involved in connecting
     multicast routing domains.

     Information useful in summary reports would include:

          Total number of systems being summarized
          Packets received and lost
          Histogram of reception quality (fraction lost)
          Histogram of receiver bandwidth
          Information on users registering

     The  total  systems  summarized number is used in order to provide for
     faster convergence times. Information on  packets  received  and  lost
     will  typically be used by network engineers looking to diagnose prob-
     lems with the MBONE. The histograms of reception quality and  receiver
     bandwidth  are propagated in order to allow senders to deduce informa-
     tion relating to the global user experience.  In order  to  allow  for

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     location  of individuals participating in a conference, the summarizer
     may wish to relay information on the users in the conference. Alterna-
     tively,  it  may  register users in a directory service via use of the
     LDAP extensions defined in [4].

     4.9.  Use of RTP translators

     Through the use of RTP translators,  it  is  possible  to  divide  RTP
     receivers  into areas in much the same way as is accomplished by OSPF.
     Through the use of summary receiver reports, information on  listener-
     ship and receiver report quality can be propagated between areas while
     reducing RTCP bandwidth usage and the RTCP sending population  on  the

     In  order  to insulate receivers within an area from external receiver
     reports, the RTP translator must  filter  external  receiver  reports,
     while  allowing  external  summary  reports and sender reports to pass
     through.  Similarly,  the  RTP  translator  will  listen  to  receiver
     reports  generated  from within its area, but will not pass them on to
     external areas. Instead, it would issue to external areas two types of
     summary  reports.  The  first will be based on the packets it receives
     and will be identical to a conventional receiver  report,  except  for
     the  use of the summary report type; the second will be a true summary
     report based on area receiver reports. It is useful to allow  receiver
     reports  from RTP translators to pass through so as to allow diagnosis
     of internal distribution  problems.  The  RTP  translator  will  allow
     internal sender reports to pass through unmodified.

     The introduction of RTP translators has several advantages:

          Improved convergence of the receiver population estimate
          Decreased RTCP bandwidth
          Improved privacy
          Listenership and reception quality information available to senders

     While  most of the above advantages are also available in the receiver
     summary approach, one advantage of the translator approach is that  it
     provides  for  greater control by the network administrator. For exam-
     ple, since summary reports would be sent  by  RTP  translators  rather
     than by client summarizers, it would be possible for administrators to
     control the propagation of user information on a  site-by-site  basis,
     rather  than  on  a per-session basis. For example, rather than having
     this sent in the summary report, it could be  stored  as  a  temporary
     attribute  in  the  area  directory server. This may be perceived as a
     substantial advantage by corporations looking  to  control  access  to
     directory  information. For those cases where it is desirable to allow
     external access to area registration information, the  IP  address  of
     the  area  directory  server  may be advertised in the summary report.
     This allows senders with appropriate privileges to retrieve conference
     registration data from the area directory server via LDAP.

     The  RTP  translator  approach  also  has several major disadvantages.
     These include:

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          Requires modifications to routers
          Increases loading on routers that now must function as RTP translators

     5.  Acknowledgements

     Thanks to Steve Casner of Precept, Mark Handley of ISI, Bob Hinden  of
     Ipsilon  and  Thomas  Pfenning,  Peter  Ford  and  Stefan  Solomon  of
     Microsoft for useful discussions of this problem space.

     6.  References

      [1]  H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson.   "RTP:  A
     Transport  Protocol  for Real-Time Applications." RFC 1889, GMD Fokus,
     January 1996.

     [2]  H. Schulzrinne.  "RTP Profile for  Audio  and  Video  Conferences
     with Minimal Control." RFC 1890, GMD Fokus, January 1996.

     [3]   M.  F.  Speer,  S.  McCanne.  "RTP Usage with Layered Multimedia
     Streams."  draft-speer-avt-layered-video-01.txt,   Sun   Microsystems,
     LBNL, June 1996.

     [4]  Y. Yaacovi, M. Wahl, K. Settle, T. Genovese.  "Lightweight Direc-
     tory Access Protocol:  Extensions  for  Dynamic  Directory  Services."
     draft-ietf-asid-ldapv3ext-02.txt,  Microsoft, Critical Angle, October,

     7.  Authors' Addresses

     Bernard Aboba
     Microsoft Corporation
     One Microsoft Way
     Redmond, WA 98052

     Phone: 206-936-6605

     Aboba                                                        [Page 10]