TDMoIP

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In computer networking and telecommunications, TDM over IP (TDMoIP) is the emulation of

time-division multiplexing (TDM) over a packet switched network (PSN). By TDM we mean a
T1, E1, T3, or E3 signal, while the PSN is based either on IP or MPLS or on raw Ethernet. A
related technology is circuit emulation, which enables transport of TDM traffic over cell-based
(ATM) networks.

TDMoIP is a type of pseudowire (PW). However, unlike other traffic types that can be carried
over pseudowires (e.g. ATM, Frame Relay, and Ethernet), TDM is a real-time bit stream, leading
to TDMoIP having unique characteristics. In addition, conventional TDM networks have
numerous special features, in particular those required in order to carry voice-grade telephony
channels. These features imply signaling systems that support a wide range of telephony
features, a rich standardization literature, and well-developed Operations and Management
(OAM) mechanisms. All of these factors must be taken into account when emulating TDM over
PSNs.

One critical issue in implementing TDM PWs is clock recovery. In native TDM networks the
physical layer carries highly accurate timing information along with the TDM data, but when
emulating TDM over PSNs this synchronization is absent. TDM timing standards can be
exacting, and conformance with these may require innovative mechanisms to adaptively
reproduce the TDM timing.

Another issue that must be addressed is TDMoIP packet loss concealment (PLC). Since TDM
data is delivered at a constant rate over a dedicated channel, the native service may have bit
errors, but data is never lost in transit. All PSNs suffer to some degree from packet loss, and this
must be compensated when delivering TDM over a PSN.

Contents
[hide]

 1 Background
 2 Handling TDM structure
 3 TDMoIP format
 4 Delay
o 4.1 Timing recovery
 5 Handling packet loss
 6 External links
[edit] Background
Communications service providers and enterprise customers are interested in deployment of
voice and leased line services over efficient Ethernet, IP and MPLS infrastructures. While Voice
over IP (VoIP) is maturing, its deployment requires an investment in new network infrastructure
and customer premises equipment (CPE). TDMoIP presents a migration path, whereby modern
packet switched networks can be used for transport, while the end-user equipment need not be
immediately replaced.

TDMoIP was first developed in 1998 by RAD Data Communications (see U.S. patent number
6,731,649) and first deployed in Sweden in 1999 by Utfors (later acquired by Telenor). Utfors
employed the first generation TDMoIP product (known as Ipmux-4) to provide bundled services
including TDM private lines, TDM leased lines and a variety of IP and Ethernet services. In
2001, the IETF set up the PWE3 working group, which was chartered to develop an architecture
for edge-to-edge pseudowires, and to produce specifications for various services, including
TDM. Other standardization forums, including the ITU and the MPLS - Frame Relay Alliance,
are also active in producing standards and implementation agreements for pseudowires.

[edit] Handling TDM structure

Although TDM can be used to carry arbitrary bit streams at the rates defined in G.702, there are
standardized methods of carrying bit streams in larger units each containing the same number of
bits, called frames. TDM framing locks the frame rate to the sampling frequency of voice traffic,
so that there are always 8000 frames per second; a T1 frame consists of 193 bits and an E1 frame
of 256 bits.

Unlike unframed TDM for which all bits are available for payload, framed TDM requires
dedicating of some number of bits per frame for synchronization and perhaps various other
functions (e.g. 1 bit per T1 frame, 8 bits per E1 frame). Framed TDM is often used to multiplex
multiple voice channels each consisting of 8000 8-bit samples per second in a sequence of
timeslots recurring in each frame. When this is done we have "channelized TDM" and additional
structure must be introduced.

In order to efficiently transport slowly varying channel associated signalling bits, second order
structures known as multiframes or superframes are defined. For example, for E1 trunks the CAS
signaling bits are updated once per multiframe of 16 frames (every 2 milliseconds) while for T1
ESF trunks the superframe is 24 frames (3 milliseconds). Other types of second order structures
are also in common use. In GSM cellular networks, the Abis channel that connects the Base
Transceiver Station (BTS) and Base Station Controller (BSC) is an E1 link with several framing
alternatives, all of which have a basic superframe duration of 20 milliseconds.

The term "structured TDM" is used to refer to TDM with any level of structure, including
'framed TDM' and 'channelized TDM'.
TDMoIP transport is denoted "structured-agnostic" when the TDM is unframed, or when it is
framed or even channelized, but the framing and channelization structure are completely
disregarded by the transport mechanisms. In such cases all structural overhead must be
transparently transported along with the payload data, and the encapsulation method employed
provides no mechanisms for its location or utilization. Structure-aware TDM transport may
explicitly safeguard TDM structure, in three conceptually distinct ways, which we shall call
structure-locking, structure-indication, and structure-reassembly.

Structure-locking ensures that packets consist of entire TDM structures or multiples/fractions

thereof. Structure-indication allows packets to contain arbitrary fragments of basic structures, but
employs pointers to indicate where the following structure commences. In structure-reassembly
components of the TDM structures may be extracted and reorganized at ingress, and the original
structure reassembled from the received constituents at egress.

[edit] TDMoIP format

TDMoIP operates by segmenting, adapting and encapsulating the TDM traffic at PSN ingress
and performing the inverse operations at PSN egress. Adaptation denotes mechanisms that
modify the payload to enable its proper restoration at the PSN egress. By using proper
adaptation, the TDM signaling and timing can be recovered, and a certain amount of packet loss
can be accommodated. Encapsulation signifies placing the adapted payload into packets of the
format required by the underlying PSN technology. For the MPLS case, ITU-T Recommendation
Y.1413 contains a complete description of the packet format.

In all cases a TDMoIP packet commences with PSN headers. These are the standard headers
used by the PSN technology, e.g. the 20-byte header of UDP/IP, or the label-stack of MPLS.
After these headers come the "PW label", a four-byte MPLS-like label that serves as to
demultiplex different TDM PWs. After the PSN header comes the four-byte TDMoIP "control
word". The control word contains a 16-bit packet sequence number (needed to detect packet re-
ordering and packet loss), payload length, and flags indicating defect conditions.

After the control word comes the TDMoIP payload. For structure-agnostic transport (SAToP)
this is simply a predetermined number of TDM octets, while for the structure-locked format the
payload is an integer number of TDM frames. For structure-indication and structure-reassembly
TDMoIP draws on proven adaptation mechanisms originally developed for ATM. A side benefit
of this choice of payload types is simplified interworking with circuit emulation services carried
over ATM networks. For statically allocated, constant bit-rate (CBR) TDM links, TDMoIP
employs ATM adaptation layer 1 (AAL1). This mechanism, defined in ITU-T standard I.363.1
and ATM Forum specification atm-vtoa-0078, was developed for carrying CBR services over
ATM. AAL1 operates by segmenting the continuous stream of TDM data into small 48-byte
cells and inserting sequencing, timing, error recovery, and synchronization information into
them. TDMoIP allows concatenation of any number of AAL1 cells into a packet (note that these
are AAL1 cells and not ATM cells, i.e. they do not include the five-byte "cell tax"). By allowing
multiple cells per packet, TDMoIP facilitates flexible tradeoffs of buffering delay (which
decreases with fewer cells per packet) for bandwidth efficiency (which increases with more cells
per packet, due to the per packet overhead). For dynamically allocated TDM links, whether the
information rate varies due to activation of time slots or due to voice activity detection, TDMoIP
employs ATM adaptation layer 2 (AAL2). This mechanism, defined in ITU-T standard I.363.2,
was developed for carrying variable bit rate (VBR) services over ATM. AAL2 operates by
buffering each TDM time slot into short minicells, inserting the time slot identifier and length
indication, sequencing, and then sending this minicell only if it carries valid information.
TDMoIP concatenates the minicells from all active time slots into a single packet. For time slots
carrying high-level data link control (HDLC) data, such as data for common channel signaling
(CCS), TDMoIP has a special adaptation that encapsulates stretches of non-idle data.

[edit] Delay
The telephony network severely constrains end-to-end delays. ITU-T G.114/G.131 states that
one-way transmission times of up to 150 ms are universally acceptable, assuming adequate echo
control is provided. These constraints are not problematic for TDM networks, where the major
component of the end-to-end delay is electrical propagation time ("light speed delay"). By
contrast, IP-based systems typically add various forms of delay, one of which is based on the
time it takes to form packets (packetization delay), which is proportional to the packet size
divided by the data rate. Packet sizes cannot be made too small or the packet header overhead
will become overwhelming. The other form of delay introduced by IP systems is the playout
delay, which needs to be added at the recipient to buffer packet delay variation and ensure a
smooth playout. VoIP systems that try to be very bandwidth efficient may also add tens of
milliseconds of algorithmic delay in the voice codec. Historically, bad implementations have
added additional, operating-system induced delays, which together with the other delays in
practice sometimes approach 100 ms even before taking propagation delays into account.

In contrast, TDMoIP maps TDM octets directly into the payload with no voice compression
algorithms and no resultant algorithmic delay. The packetization latency added by TDMoIP
depends on the number of cells per packet but is typically in the single millisecond range due to
the higher data rate of a complete multiplex as compared to a single VoIP flow. Playout delay
considerations do not differ materially between TDMoIP and VoIP, however, so both work best
on paths with controlled packet delay variation (strong overprovisioning or "QoS").

[edit] Timing recovery

Native TDM networks rely on hierarchical distribution of timing. Somewhere in the network
there is at least one extremely accurate primary reference clock with a long-term accuracy of 1 x
10^-11. This node, which offers Stratum 1 accuracy, provides the reference clock to secondary
nodes with Stratum 2 accuracy. The secondary nodes then provide a time reference to Stratum 3
nodes. This hierarchy of time synchronization is essential for the proper functioning of the
network as a whole.

Packets in the PSN reach their destination with delay that has a random component, known as
packet delay variation (PDV). When emulating TDM transport on such a network, this
randomness may be overcome by placing the TDM packets into a jitter buffer from which data
can be read out at a constant rate for delivery to TDM end-user equipment. The problem is that
the TDM source time reference is no longer available, and the precise rate at which the data are
to be "clocked out" of the jitter buffer is unknown.

In certain cases timing may be derived from the TDM equipment at both ends of the PW. Since
each of these clocks is highly accurate, they necessarily agree to high order. The problem arises
when at most one side of the TDMoIP tunnel has a highly accurate time standard. For ATM
networks, which define a physical layer that carries timing, the synchronous residual time stamp
(SRTS) method may be used; IP/MPLS networks, however, do not define the physical layer and
thus cannot specify the accuracy of its clock.

Hence, in many cases the only alternative is to attempt to recover the clock based exclusively on
the TDMoIP traffic, a technology known as "adaptive clock recovery". This is possible since the
source TDM device is producing bits at a constant rate determined by its clock, although this rate
is hidden by the PDV. The task of clock recovery is thus an "averaging" process that negates the
effect of the random PDV and captures the average rate of transmission of the original bit
stream.

[edit] Handling packet loss

While proper application of traffic engineering and quality-of-service (QoS) is expected to
minimize packet loss, packets will at times arrive at the egress out of order. They may also have
been dropped altogether within the PSN. The TDMoIP control word described above includes a
16-bit sequence number for detecting and handling lost and mis-ordered packets. In the case of
lost packets, TDMoIP requires insertion of interpolation packets to maintain TDM timing.
Misordered packets may be either reordered or dropped and interpolated.

While the insertion of arbitrary packets may be sufficient to maintain the TDM timing, in voice
applications packet loss can cause gaps or errors that result in choppy, annoying, or even
unintelligible speech. The precise effect of packet loss on voice quality and the development of
packet loss concealment algorithms have been the subject of detailed study in the VoIP
community, but their results are not directly applicable to the TDMoIP case. This is because
VoIP packets typically contain between 80 samples (10 ms) and 240 samples (30 ms) of the
speech signal, while TDMoIP packets may contain only a small number of samples. Since
TDMoIP packets are so small, it is acceptable to simply insert a constant value in place of any
lost speech samples. Assuming that the input signal is zero-mean (i.e. contains no DC
component), minimal distortion is attained when this constant is set to zero. Alternatively, more
sophisticated approaches call for optimally predicting the values of missing samples.