RFC 7893 Pseudowire Congestion Considerations

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INFORMATIONAL

Internet Engineering Task Force (IETF)                        Y(J) Stein
Request for Comments: 7893                       RAD Data Communications
Category: Informational                                         D. Black
ISSN: 2070-1721                                          EMC Corporation
                                                              B. Briscoe
                                                                      BT
                                                               June 2016


                  Pseudowire Congestion Considerations

Abstract

   Pseudowires (PWs) have become a common mechanism for tunneling
   traffic and may be found in unmanaged scenarios competing for network
   resources both with other PWs and with non-PW traffic, such as TCP/IP
   flows.  Thus, it is worthwhile specifying under what conditions such
   competition is acceptable, i.e., the PW traffic does not
   significantly harm other traffic or contribute more than it should to
   congestion.  We conclude that PWs transporting responsive traffic
   behave as desired without the need for additional mechanisms.  For
   inelastic PWs (such as Time Division Multiplexing (TDM) PWs), we
   derive a bound under which such PWs consume no more network capacity
   than a TCP flow.  For TDM PWs, we find that the level of congestion
   at which the PW can no longer deliver acceptable TDM service is never
   significantly greater, and is typically much lower, than this bound.
   Therefore, as long as the PW is shut down when it can no longer
   deliver acceptable TDM service, it will never do significantly more
   harm than even a single TCP flow.  If the TDM service does not
   automatically shut down, a mechanism to block persistently
   unacceptable TDM pseudowires is required.

Status of This Memo

   This document is not an Internet Standards Track specification; it is
   published for informational purposes.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Not all documents
   approved by the IESG are a candidate for any level of Internet
   Standard; see Section 2 of RFC 7841.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   http://www.rfc-editor.org/info/rfc7893.




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Copyright Notice

   Copyright (c) 2016 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   5
   3.  PWs Comprising Elastic Flows  . . . . . . . . . . . . . . . .   6
   4.  PWs Comprising Inelastic Flows  . . . . . . . . . . . . . . .   7
   5.  Conclusions . . . . . . . . . . . . . . . . . . . . . . . . .  19
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .  19
   7.  Informative References  . . . . . . . . . . . . . . . . . . .  19
   Appendix A.  Loss Probabilities for TDM PWs . . . . . . . . . . .  22
   Appendix B.  Effect of Packet Loss on Voice Quality for
                Structure-Aware TDM PWs  . . . . . . . . . . . . . .  23
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  27























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1.  Introduction

   A pseudowire (PW) (see [RFC3985]) is a construct for tunneling a
   native service, such as Ethernet or TDM, over a Packet Switched
   Network (PSN), such as IPv4, IPv6, or MPLS.  The PW packet
   encapsulates a unit of native service information by prepending the
   headers required for transport in the particular PSN (which must
   include a demultiplexer field to distinguish the different PWs) and
   preferably the 4-byte Pseudowire Emulation Edge-to-Edge (PWE3)
   control word.

   PWs have no bandwidth reservation or control mechanisms, meaning that
   when multiple PWs are transported in parallel, and/or in parallel
   with other flows, there is no defined means for allocating resources
   for any particular PW, or for preventing the negative impact of a
   particular PW on neighboring flows.  The case where the service
   provider network provisions a PW with sufficient capacity is well
   understood and will not be discussed further here.  Concerns arise
   when PWs share network capacity with elastic or congestion-responsive
   traffic, whether that capacity sharing was planned by a service
   provider or results from PW deployment by an end user.

   PWs are most often placed in MPLS tunnels, but we herein restrict
   ourselves to PWs in IPv4 or IPv6 PSNs; MPLS PSNs are beyond the scope
   of this document.  There are several mechanisms that enable
   transporting PWs over an IP infrastructure, including:

   o  UDP/IP encapsulations as defined for TDM PWs [RFC4553] [RFC5086]
      [RFC5087],

   o  PWs based on Layer 2 Tunneling Protocol (L2TPv3) [RFC3931],

   o  MPLS PWs directly over IP according to RFC 4023 [RFC4023], and

   o  MPLS PWs over Generic Routing Encapsulation (GRE) over IP
      according to RFC 4023 [RFC4023].

   Whenever PWs are transported over IP, they may compete for network
   resources with neighboring congestion-responsive flows (e.g., TCP
   flows).  In this document, we study the effect of PWs on such
   neighboring flows, and discover that the negative impact of PW
   traffic is generally no worse than that of congestion-responsive
   flows [RFC2914] [RFC5033].

   At first glance, one may consider a PW transported over IP to be
   considered as a single flow, on par with a single TCP flow.  Were we
   to accept this tenet, we would require a PW to back off under
   congestion to consume no more bandwidth than a single TCP flow under



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   such conditions (see [RFC5348]).  However, since PWs may carry
   traffic from many users, it makes more sense to consider each PW to
   be equivalent to multiple TCP flows.

   The following two sections consider PWs of two types:

   Elastic Flows:
      Section 3 concludes that the response to congestion of a PW
      carrying elastic (e.g., TCP) flows is no different from the
      aggregated behaviors of the individual elastic flows, had they not
      been encapsulated within a PW.

   Inelastic Flows:
      Section 4 considers the case of inelastic constant bit rate (CBR)
      TDM PWs [RFC4553] [RFC5086] [RFC5087] competing with TCP flows.
      Such PWs require a preset amount of bandwidth, that may be lower
      or higher than that consumed by an otherwise unconstrained TCP
      flow under the same network conditions.  In any case, such a PW is
      unable to respond to congestion in a TCP-like manner; although
      admittedly the total bandwidth it consumes remains constant and
      does not increase to consume additional bandwidth as TCP rates
      back off.  For TDM services, we will show that TDM service quality
      degradation generally occurs before the TDM PW becomes TCP-
      unfriendly.  For TDM services that do not automatically shut down
      when they persistently fail to comply with acceptable TDM service
      criteria, a transport circuit breaker [CIRCUIT-BREAKER] may be
      employed as a last resort to shut down a TDM pseudowire that can
      no longer deliver acceptable service.

   Thus, in both cases, pseudowires will not inflict significant harm on
   neighboring TCP flows, as in one case they respond adequately to
   congestion, and in the other they would be shut down due to being
   unable to deliver acceptable service before harming neighboring
   flows.

   Note: This document contains a large number of graphs that are
   necessary for its understanding, but could not be rendered in ASCII.
   It is strongly suggested that the PDF version be consulted.













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2.  Terminology

   The following acronyms are used in this document:

   AIS     Alarm Indication Signal (see [G775])

   BER     Bit Error Rate [G826]

   BW      Bandwidth

   CBR     Constant Bit Rate

   ES      Errored Second [G826]

   ESR     Errored Second Rate [G826]

   GRE     Generic Routing Encapsulation [RFC2784]

   L2TPv3  Layer 2 Tunneling Protocol Version 3 [RFC3931]

   MOS     Mean Opinion Score [P800]

   MPLS    Multiprotocol Label Switching [RFC3031]

   NSP     Native Service Processing [RFC3985]

   PLR     Packet Loss Ratio

   PSN     Packet Switched Network [RFC3985]

   PW      Pseudowire [RFC3985]

   SAToP   Structure-Agnostic TDM over Packet [RFC4553]

   SES     Severely Errored Seconds [G826]

   SESR    Severely Errored Seconds Ratio [G826]

   TCP     Transmission Control Protocol

   TDM     Time Division Multiplexing [G703]

   UDP     User Datagram Protocol








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3.  PWs Comprising Elastic Flows

   In this section, we consider Ethernet PWs that primarily carry
   congestion-responsive traffic.  We expand on the remark in Section 8
   (Congestion Control) of [RFC4553], and show that the desired
   congestion avoidance behavior is automatically obtained and
   additional mechanisms are not needed.

   Let us assume that an Ethernet PW aggregating several TCP flows is
   flowing alongside several TCP/IP flows.  Each Ethernet PW packet
   carries a single Ethernet frame that carries a single IP packet that
   carries a single TCP segment.  Thus, if congestion is signaled by an
   intermediate router dropping a packet, a single end-user TCP/IP
   packet is dropped, whether or not that packet is encapsulated in the
   PW.

   The result is that the individual TCP flows inside the PW experience
   the same drop probability as the non-PW TCP flows.  Thus, the
   behavior of a TCP sender (retransmitting the packet and appropriately
   reducing its sending rate) is the same for flows directly over IP and
   for flows inside the PW.  In other words, individual TCP flows are
   neither rewarded nor penalized for being carried over the PW.  An
   elastic PW does not behave as a single TCP flow, as it will consume
   the aggregated bandwidth of its component flows; yet if its component
   TCP flows backs off by some percentage, the bandwidth of the PW as a
   whole will be reduced by the very same percentage, purely due to the
   combined effect of its component flows.

   This is, of course, precisely the desired behavior.  Were individual
   TCP flows rewarded for being carried over a PW, this would create an
   incentive to create PWs for no operational reason.  Were individual
   flows penalized, there would be a deterrence that could impede
   pseudowire deployment.

   There have been proposals to add additional TCP-friendly mechanisms
   to PWs, for example by carrying PWs over DCCP.  In light of the above
   arguments, it is clear that this would force the PW down to the
   bandwidth of a single flow, rather than N flows, and penalize the
   constituent TCP flows.  In addition, the individual TCP flows would
   still back off due to their endpoints being oblivious to the fact
   that they are carried over a PW.  This would further degrade the
   flow's throughput as compared to a non-PW-encapsulated flow, in
   contradiction to desirable behavior.








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   We have limited our treatment to the case of TCP traffic carried by
   Ethernet PWs (which are by far the most commonly deployed packet-
   carrying pseudowires), but it is not overly difficult to show that
   our result is equally valid for other PW types, such as ATM or frame-
   relay pseudowires.

4.  PWs Comprising Inelastic Flows

   Inelastic PWs, such as TDM PWs [RFC4553] [RFC5086] [RFC5087], are
   potentially more problematic than the elastic PWs of the previous
   section.  As mentioned in Section 8 (Congestion Control) of
   [RFC4553], being constant bit rate (CBR), TDM PWs can't incrementally
   respond to congestion in a TCP-like fashion.  On the other hand,
   being CBR, TDM PWs do not make things worse by attempting to capture
   additional bandwidth when neighboring TCP flows back off.

   Since a TDM PW consumes a constant amount of bandwidth, if the
   bandwidth occupied by a TDM PW endangers the network as a whole, it
   might seem that the only recourse is to shut it down, denying service
   to all customers of the TDM native service.  Nonetheless, under
   certain conditions it may be possible to reduce the bandwidth
   consumption of an emulated TDM service.  A prevalent case is that of
   a TDM native service that carries voice channels that may not all be
   active.  The ATM Adaptation Layer 2 (AAL2) mode of [RFC5087] (perhaps
   along with connection admission control) can enable bandwidth
   adaptation, at the expense of more sophisticated native service
   processing (NSP).

   In the following, we will focus on structure-agnostic TDM PWs
   [RFC4553] although similar analysis can be readily applied to
   structure-aware PWs (see Appendix B).  We will show that, for many
   cases of interest, a TDM PW, even when treated as a single flow, will
   behave in a reasonable manner without any additional mechanisms.  We
   also show that, at the level of congestion when a TDM PW can no
   longer deliver acceptable TDM service, a single unconstrained TCP
   flow would typically still consume more capacity than a whole TDM PW.
   Therefore, to ensure that a TDM PW does not inflict significantly
   more harm than a TCP flow, it suffices to shut down a TDM PW that is
   persistently unable to deliver acceptable TDM service.  This shutting
   down could be accomplished by employing a managed transport circuit
   breaker, by which we mean an automatic mechanism for terminating an
   unresponsive flow during persistently high levels of congestion
   [CIRCUIT-BREAKER].  Note that a transport circuit breaker is intended
   as a protection mechanism of last resort, just as an electrical
   circuit breaker is only triggered when absolutely necessary.






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   For the avoidance of doubt, the above does not say that a TDM PW
   should be shut down when it becomes TCP-unfriendly.  It merely says
   that the act of shutting down a TDM PW that can no longer deliver
   acceptable TDM service ensures that the PW does not contribute to
   congestion significantly more than a TCP flow would.  Also, note that
   being unable to deliver acceptable TDM service for a short amount of
   time is insufficient justification for shutting down a TDM PW.  While
   TCP flows react within a round-trip time, service commissioning and
   decommissioning are generally time-consuming processes that should
   only be undertaken when it becomes clear that the congestion is not
   transient.

   In order to quantitatively compare TDM PWs to TCP flows, we will
   compare the effect of TDM PW traffic with that of TCP traffic having
   the same packet size and delay.  This is potentially an overly
   pessimistic comparison, as TDM PW packets are frequently configured
   to be short in order to minimize latency, while TCP packets are free
   to be much larger.

   There are two network parameters relevant to our discussion, namely
   the one-way delay (D) and the packet loss ratio (PLR).  The one-way
   delay of a native TDM service consists of the physical time-of-flight
   plus 125 microseconds for each TDM switch traversed, and is thus very
   small as compared to typical PSN network-crossing latencies.  Since
   TDM services are designed with this low latency in mind, emulated TDM
   services are usually required to have similar low end-to-end delay.
   In our comparisons, we will only consider one-way delays of a few
   milliseconds.

   Regarding packet loss, the relevant RFCs specify actions to be
   carried out upon detecting a lost packet.  Structure-agnostic
   transport has no alternative to outputting an "all-ones" Alarm
   Indication Signal (AIS) pattern towards the TDM circuit, which, when
   long enough in duration, is recognized by the receiving TDM device as
   a fault indication (see Appendix A).  TDM standards (such as [G826])
   place stringent limits on the number of such faults tolerated.
   Calculations presented in Appendix A show that only loss
   probabilities in the realm of fractions of a percent are relevant for
   structure-agnostic transport.  Structure-aware transport regenerates
   frame alignment signals, thus avoiding AIS indications resulting from
   infrequent packet loss.  Furthermore, for TDM circuits carrying voice
   channels, the use of packet loss concealment algorithms is possible
   (such algorithms have been previously described for TDM PWs).
   However, even structure-aware transport ceases to provide a useful
   service at about 2 percent loss probability.  Hence, in our
   comparisons we will only consider PLRs of 1 or 2 percent.





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   TCP Friendly Rate Control (TFRC) [RFC5348] provides a simplified
   formula for TCP throughput as a function of round-trip delay and
   packet loss ratio.

                                    S
       X     = ------------------------------------------------
                 R  ( sqrt(2p/3) + 12 sqrt(3p/8) p (1+32p^2) )

   where:

      X is the average sending rate in bytes per second,

      S is the segment (packet payload) size in bytes,

      R is the round-trip time in seconds,

      p is the packet loss probability (i.e., PLR/100).

   We can now compare the bandwidth consumed by TDM pseudowires with
   that of a TCP flow for a given packet loss ratio and one-way end-to-
   end delay (taken to be half the round-trip delay R).  The results are
   depicted in the accompanying figures (available only in the PDF
   version of this document).  In Figures 1 and 2, we see the
   conventional rate vs. packet loss plot for low-rate TDM (both T1 and
   E1) traffic, as well as TCP traffic with the same payload size (64 or
   256 bytes respectively).  Since the TDM rates are constant (T1 and E1
   having payload throughputs of 1.544 Mbps and 2.048 Mbps
   respectively), and Structure-Agnostic TDM over packet (SAToP) can
   only faithfully emulate a TDM service up to a PLR of about half a
   percent, the T1 and E1 pseudowires occupy line segments on the graph.
   On the other hand, the TCP rate equation produces rate curves
   dependent on both one-way delay and packet loss.

   For large packet sizes, short one-way delays, and low packet loss
   ratios, the TDM pseudowires typically consume much less bandwidth
   than TCP would under identical conditions.  For small packets, long
   one-way delays, and high packet loss ratios, TDM PWs potentially
   consume more bandwidth, but only marginally.  Furthermore, our
   "apples to apples" comparison forced the TCP traffic to use packets
   of sizes smaller than would be typical.

   Similarly, in Figures 3 and 4 we repeat the exercise for higher rate
   E3 and T3 (rates 34.368 and 44.736 Mbps respectively) pseudowires,
   allowing delays and PLRs suitable for these signals.  We see that the
   TDM pseudowires consume much less bandwidth than TCP, for all
   reasonable parameter combinations.





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   --------------------------------------------------------------------
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             Figure 1: E1/T1 PWs vs. TCP for Segment Size 64B
































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             Figure 2: E1/T1 PWs vs. TCP for Segment Size 256B
































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             Figure 3: E3/T3 PWs vs. TCP for Segment Size 536B
































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            Figure 4: E3/T3 PWs vs. TCP for Segment Size 1024B
































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   We can use the TCP rate equation to determine the precise conditions
   under which a TDM PW consumes no more bandwidth than a TCP flow
   between the same endpoints under identical conditions.  Replacing the
   round-trip delay with twice the one-way delay D, setting the
   bandwidth to that of the TDM service BW, and the segment size to be
   the TDM fragment (taking into account the PWE3 control word), we
   obtain the following condition for a TDM PW:

              4 S
       D < -----------
             BW f(p)

   where:

      D is the one-way delay,

      S is the TDM segment size (packet excluding overhead) in bytes,

      BW is the TDM service bandwidth in bits per second,

      f(p) = sqrt(2p/3) + 12 sqrt(3p/8) p (1+32p^2).

   One may view this condition as defining a "friendly" operating
   envelope for a TDM PW, as a TDM PW that occupies no more bandwidth
   than a TCP flow causes no more congestion than that TCP flow.  Under
   this condition, it is acceptable to place the TDM PW alongside
   congestion-responsive traffic such as TCP.  On the other hand, were
   the TDM PW to consume significantly more bandwidth than a TCP flow,
   it could contribute disproportionately to congestion, and its mixture
   with congestion-responsive traffic might be inappropriate.  Note that
   we are sidestepping any debate over the validity of the TCP-
   friendliness concept and merely saying that there can be no question
   that a TDM PW is acceptable if it causes no more congestion than a
   single TCP flow.

   We derived this condition assuming steady-state conditions, and thus
   two caveats are in order.  First, the condition does not specify how
   to treat a TDM PW that initially satisfies the condition, but is then
   faced with a deteriorating network environment.  In such cases, one
   additionally needs to analyze the reaction times of the responsive
   flows to congestion events.  Second, the derivation assumed that the
   TDM PW was competing with long-lived TCP flows, because under this
   assumption it was straightforward to obtain a quantitative comparison
   with something widely considered to offer a safe response to
   congestion.  Short-lived TCP flows may find themselves disadvantaged
   as compared to a long-lived TDM PW satisfying the above condition.





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   We see in Figures 5 and 6 that TDM pseudowires carrying T1 or E1
   native services satisfy the condition for all parameters of interest
   for large packet sizes (e.g., S=512 bytes of TDM data).  For the
   SAToP default of 256 bytes, as long as the one-way delay is less than
   10 milliseconds, the loss probability can exceed 0.3 or 0.6 percent.
   For packets containing 128 or 64 bytes, the constraints are more
   troublesome, but there are still parameter ranges where the TDM PW
   consumes less than a TCP flow under similar conditions.  Similarly,
   Figures 7 and 8 demonstrate that E3 and T3 native services with the
   SAToP default of 1024 bytes of TDM per packet satisfy the condition
   for a broad spectrum of delays and PLRs.

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              Figure 5: TCP Compatibility Areas for T1 SAToP




















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              Figure 6: TCP Compatibility Areas for E1 SAToP
































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              Figure 7: TCP Compatibility Areas for E3 SAToP
































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              Figure 8: TCP Compatibility Areas for T3 SAToP
































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5.  Conclusions

   The figures presented in the previous section demonstrate that TDM
   service quality degradation generally occurs before the TDM PW would
   consume more bandwidth than a comparable TCP flow.  Thus, while TDM
   PWs are unable to respond to congestion in a TCP-like fashion, TDM
   PWs that are able to deliver acceptable TDM service do not contribute
   to congestion significantly more than a TCP flow.

   Combined with our earlier determination that Ethernet PWs
   automatically respond in a TCP-like fashion (see Section 3), our
   final conclusion is that PW-specific congestion-avoidance mechanisms
   are generally not required.  This is true even for TDM PWs, assuming
   that the TDM management plane initiates service shutdown when service
   parameters are persistently below levels required by the relevant TDM
   standards.  If the TDM service does not automatically shut down, a
   mechanism to block persistently unacceptable TDM pseudowires is
   required, or a transport circuit breaker [CIRCUIT-BREAKER] may be
   triggered as a last resort.

6.  Security Considerations

   This document does not introduce any new congestion-specific
   mechanisms and thus does not introduce any new security
   considerations above those present for PWs in general.

7.  Informative References

   [CIRCUIT-BREAKER]
              Fairhurst, G., "Network Transport Circuit Breakers", Work
              in Progress, draft-ietf-tsvwg-circuit-breaker-15, April
              2016.

   [G703]     ITU-T, "Physical/electrical characteristics of
              hierarchical digital interfaces", ITU Recommendation
              G.703, April 2016.

   [G775]     ITU-T, "Loss of Signal (LOS), Alarm Indication Signal
              (AIS) and Remote Defect Indication (RDI) defect detection
              and clearance criteria for PDH signals",
              ITU Recommendation G.775, October 1998.

   [G826]     ITU-T, "Error Performance Parameters and Objectives for
              International Constant Bit Rate Digital Paths at or above
              Primary Rate", ITU Recommendation G.826, December 2002.






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   [P50App1]  ITU-T, "Telephone Transmission Quality, Telephone
              Installations, Local Line Networks: Appendix 1",
              ITU-T Recommendation P.50, February 1998.

   [P800]     ITU-T, "Methods for subjective determination of
              transmission quality", ITU Recommendation P.800, June
              1998.

   [P862]     ITU-T, "Perceptual evaluation of speech quality (PESQ): An
              objective method for end-to-end speech quality assessment
              of narrow-band telephone networks and speech codecs",
              ITU Recommendation P.826, February 2001.

   [PACKET-LOSS]
              Stein, J(Y). and I. Druker, "The Effect of Packet Loss on
              Voice Quality for TDM over Pseudowires", Work in
              Progress, draft-stein-pwe3-tdm-packetloss-01, December
              2003.

   [RFC2784]  Farinacci, D., Li, T., Hanks, S., Meyer, D., and P.
              Traina, "Generic Routing Encapsulation (GRE)", RFC 2784,
              DOI 10.17487/RFC2784, March 2000,
              <http://www.rfc-editor.org/info/rfc2784>.

   [RFC2914]  Floyd, S., "Congestion Control Principles", BCP 41,
              RFC 2914, DOI 10.17487/RFC2914, September 2000,
              <http://www.rfc-editor.org/info/rfc2914>.

   [RFC3031]  Rosen, E., Viswanathan, A., and R. Callon, "Multiprotocol
              Label Switching Architecture", RFC 3031,
              DOI 10.17487/RFC3031, January 2001,
              <http://www.rfc-editor.org/info/rfc3031>.

   [RFC3931]  Lau, J., Ed., Townsley, M., Ed., and I. Goyret, Ed.,
              "Layer Two Tunneling Protocol - Version 3 (L2TPv3)",
              RFC 3931, DOI 10.17487/RFC3931, March 2005,
              <http://www.rfc-editor.org/info/rfc3931>.

   [RFC3985]  Bryant, S., Ed. and P. Pate, Ed., "Pseudo Wire Emulation
              Edge-to-Edge (PWE3) Architecture", RFC 3985,
              DOI 10.17487/RFC3985, March 2005,
              <http://www.rfc-editor.org/info/rfc3985>.

   [RFC4023]  Worster, T., Rekhter, Y., and E. Rosen, Ed.,
              "Encapsulating MPLS in IP or Generic Routing Encapsulation
              (GRE)", RFC 4023, DOI 10.17487/RFC4023, March 2005,
              <http://www.rfc-editor.org/info/rfc4023>.




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   [RFC4553]  Vainshtein, A., Ed. and YJ. Stein, Ed., "Structure-
              Agnostic Time Division Multiplexing (TDM) over Packet
              (SAToP)", RFC 4553, DOI 10.17487/RFC4553, June 2006,
              <http://www.rfc-editor.org/info/rfc4553>.

   [RFC5033]  Floyd, S. and M. Allman, "Specifying New Congestion
              Control Algorithms", BCP 133, RFC 5033,
              DOI 10.17487/RFC5033, August 2007,
              <http://www.rfc-editor.org/info/rfc5033>.

   [RFC5086]  Vainshtein, A., Ed., Sasson, I., Metz, E., Frost, T., and
              P. Pate, "Structure-Aware Time Division Multiplexed (TDM)
              Circuit Emulation Service over Packet Switched Network
              (CESoPSN)", RFC 5086, DOI 10.17487/RFC5086, December 2007,
              <http://www.rfc-editor.org/info/rfc5086>.

   [RFC5087]  Stein, Y(J)., Shashoua, R., Insler, R., and M. Anavi,
              "Time Division Multiplexing over IP (TDMoIP)", RFC 5087,
              DOI 10.17487/RFC5087, December 2007,
              <http://www.rfc-editor.org/info/rfc5087>.

   [RFC5348]  Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
              Friendly Rate Control (TFRC): Protocol Specification",
              RFC 5348, DOI 10.17487/RFC5348, September 2008,
              <http://www.rfc-editor.org/info/rfc5348>.


























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Appendix A.  Loss Probabilities for TDM PWs

   ITU-T Recommendation G.826 [G826] specifies limits on the Errored
   Second Ratio (ESR) and the Severely Errored Second Ratio (SESR).  For
   our purposes, we will simplify the definitions and understand an
   Errored Second (ES) to be a second of time during which a TDM bit
   error occurred or a defect indication was detected.  A Severely
   Errored Second (SES) is an ES second during which the Bit Error Rate
   (BER) exceeded one in one thousand (10^-3).  Note that if the error
   condition AIS was detected according to the criteria of ITU-T
   Recommendation G.775 [G775], an SES was considered to have occurred.
   The respective ratios are the fraction of ES or SES to the total
   number of seconds in the measurement interval.

   All TDM signals run at 8000 frames per second (higher rate TDM
   signals have longer frames).  So, assuming an integer number of TDM
   frames per TDM PW packet, the number of packets per second is given
   by packets per second = 8000 / (frames per packet).  Prevalent cases
   are 1, 2, 4, and 8 frames per packet, translating to 8000, 4000,
   2000, and 1000 packets per second, respectively.

   For both E1 and T1 TDM circuits, G.826 allows an ESR of 4% (0.04),
   and an SESR of 0.2% (0.002).  For E3 and T3, the ESR must be no more
   than 7.5% (0.075), while the SESR is unchanged.  Focusing on E1
   circuits, the ESR of 4% translates (assuming the worst case of
   isolated exactly periodic packet loss) to a packet loss event no more
   than every 25 seconds.  However, once a packet is lost, another
   packet lost in the same second doesn't change the ESR, although it
   may contribute to the ES becoming an SES.  Thus for 1, 2, 4, and 8
   frames per packet, the maximum allowed packet loss probability is
   0.0005%, 0.001%, 0.002%, and 0.004% respectively.

   These extremely low allowed packet loss probabilities are only for
   the worst case scenario.  With tail-drop buffers, when packet loss is
   above 0.001%, it is likely that loss bursts will occur.  If the lost
   packets are sufficiently close together (we ignore the precise
   details here), then the permitted packet loss ratio increases by the
   appropriate factor, without G.826 being cognizant of any change.
   Hence, the worst-case analysis is expected to be extremely
   pessimistic for real networks.  Next, we will consider the opposite
   extreme and assume that all packet loss events are in periodic loss
   bursts.  In order to minimize the ESR, we will assume that the burst
   lasts no more than one second, and so we can afford to lose in each
   burst no more than the number of packets transmitted in one second.
   As long as such one-second bursts do not exceed four percent of the
   time, we still maintain the allowable ESR.  Hence, the maximum





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   permissible packet loss ratio is 4%.  Of course, this estimate is
   extremely optimistic, and furthermore does not take into
   consideration the SESR criteria.

   As previously explained, an SES is declared whenever AIS is detected.
   There is a major difference between structure-aware and structure-
   agnostic transport in this regards.  When a packet is lost, SAToP
   outputs an "all-ones" pattern to the TDM circuit, which is
   interpreted as AIS according to G.775 [G775].  For E1 circuits, G.775
   specifies that AIS is detected when four consecutive TDM frames have
   no more than 2 alternations.  This means that if a PW packet or
   consecutive packets containing at least four frames are lost, and
   four or more frames of "all-ones" output to the TDM circuit, an SES
   will be declared.  Thus burst packet loss, or packets containing a
   large number of TDM frames, lead SAToP to cause high SESR, which is
   20 times more restricted than ESR.  On the other hand, since
   structure-aware transport regenerates the correct frame alignment
   pattern, even when the corresponding packet has been lost, packet
   loss will not cause declaration of SES.  This is the main reason that
   SAToP is much more vulnerable to packet loss than the structure-aware
   methods.

   For realistic networks, the maximum allowed packet loss for SAToP
   will be intermediate between the extremely pessimistic estimates and
   the extremely optimistic ones.  In order to numerically gauge the
   situation, we have modeled the network as a four-state Markov model,
   (corresponding to a successfully received packet, a packet received
   within a loss burst, a packet lost within a burst, and a packet lost
   when not within a burst).  This model is an extension of the widely
   used Gilbert model.  We set the transition probabilities in order to
   roughly correspond to anecdotal evidence, namely low background
   isolated packet loss, and infrequent bursts wherein most packets are
   lost.  Such simulation shows that up to 0.5% average packet loss may
   occur and the recovered TDM still conforms to the G.826 ESR and SESR
   criteria.

Appendix B.  Effect of Packet Loss on Voice Quality for Structure-Aware
             TDM PWs

   Packet loss in voice traffic causes audio artifacts such as choppy,
   annoying, or even unintelligible speech.  The precise effect of
   packet loss on voice quality has been the subject of detailed study
   in the Voice over IP (VoIP) community, but VoIP results are not
   directly applicable to TDM PWs.  This is because VoIP packets
   typically contain over 10 milliseconds of the speech signal, while
   multichannel TDM packets may contain only a single sample, or perhaps
   a very small number of samples.




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   The effect of packet loss on TDM PWs has been previously reported
   [PACKET-LOSS].  In that study, it was assumed that each packet
   carried a single sample of each TDM timeslot (although the extension
   to multiple samples is relatively straightforward and does not
   drastically change the results).  Four sample replacement algorithms
   were compared, differing in the value used to replace the lost
   sample:

   1.  Replacing every lost sample by a preselected constant (e.g., zero
       or "AIS" insertion).

   2.  Replacing a lost sample by the previous sample.

   3.  Replacing a lost sample by linear interpolation between the
       previous and following samples.

   4.  Replacing the lost sample by STatistically Enhanced INterpolation
       (STEIN).

   Only the first method is applicable to SAToP transport, as structure
   awareness is required in order to identify the individual voice
   channels.  For structure-aware transport, the loss of a packet is
   typically identified by the receipt of the following packet, and thus
   the following sample is usually available.  The last algorithm posits
   the Linear-Predictive Coding (LPC) speech generation model and
   derives lost samples based on available samples both before and after
   each lost sample.

   The four algorithms were compared in a controlled experiment in which
   speech data was selected from English and American English subsets of
   the ITU-T P.50 Appendix 1 corpus [P50App1] and consisted of 16
   speakers, eight male and eight female.  Each speaker spoke either
   three or four sentences, for a total of between seven and 15 seconds.
   The selected files were filtered to telephony quality using modified
   IRS filtering and down-sampled to 8 kHz.  Packet loss of 0, 0.25,
   0.5, 0.75, 1, 2, 3, 4, and 5 percent were simulated using a uniform
   random number generator (bursty packet loss was also simulated but is
   not reported here).  For each file, the four methods of lost sample
   replacement were applied and the Mean Opinion Score (MOS) was
   estimated using PESQ [P862].  Figure 9 depicts the PESQ-derived MOS
   for each of the four replacement methods for packet drop
   probabilities up to 5%.









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   --------------------------------------------------------------------
   I                                                                  I
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   I     PESQ-MOS as a function of packet drop probability            I
   I                                                                  I
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   I                     (only in PDF version)                        I
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    Figure 9: PESQ-Derived MOS as a Function of Packet-Drop Probability

   For all cases, the MOS resulting from the use of zero insertion is
   less than that obtained by replacing with the previous sample, which
   in turn is less than that of linear interpolation, which is slightly
   less than that obtained by statistical interpolation.

   Unlike the artifacts that speech compression methods may produce when
   subject to buffer loss, packet loss here effectively produces
   additive white impulse noise.  The subjective impression is that of
   static noise on AM radio stations or crackling on old phonograph
   records.  For a given PESQ-derived MOS, this type of degradation is
   more acceptable to listeners than choppiness or tones common in VoIP.

   If MOS>4 (full toll quality) is required, then the following packet
   drop probabilities are allowable:

      zero insertion - 0.05%

      previous sample - 0.25%

      linear interpolation - 0.75%

      STEIN - 2%





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   If MOS>3.75 (barely perceptible quality degradation) is acceptable,
   then the following packet drop probabilities are allowable:

      zero insertion - 0.1%

      previous sample - 0.75%

      linear interpolation - 3%

      STEIN - 6.5%

   If MOS>3.5 (cell phone quality) is tolerable, then the following
   packet drop probabilities are allowable:

      zero insertion - 0.4%

      previous sample - 2%

      linear interpolation - 8%

      STEIN - 14%






























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Authors' Addresses

   Yaakov (Jonathan) Stein
   RAD Data Communications
   24 Raoul Wallenberg St., Bldg C
   Tel Aviv  69719
   Israel

   Phone: +972 (0)3 645-5389
   Email: yaakov_s@rad.com


   David L. Black
   EMC Corporation
   176 South St.
   Hopkinton, MA  69719
   United States

   Phone: +1 (508) 293-7953
   Email: david.black@emc.com


   Bob Briscoe
   BT

   Email: ietf@bobbriscoe.net
   URI:   http://bobbriscoe.net/
























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