Voice Transport Techniques over MPLS

Junaid Ahmed Zubairi, Ph.D., Member, IEEE

with DiffServ for prioritized handling of the QoS traffic. ITU-

Abstract- In this paper, we discuss currently evolving
technologies for packet based real-time voice communication.
Packet based voice has gained popularity in the toll-quality
market due to its flexibility and cost effectiveness. IMS (IP
Multimedia Subsystem) is being developed for converged voice
and data services. While initially it was intended for mobile
networks, it has been expanded to include support for fixed
networks. In the wired network, the quality of service needed for
media traffic can be provided by MPLS and related protocols.
The traffic engineering and quality of service mechanisms of
MPLS and Diffserv allow the differential treatment of premium
traffic on predefined LSPs (Label Switched Path). Recently T-
MPLS has been introduced as the transport network for
connection oriented packet switched traffic, dropping some of the
MPLS features that were irrelevant for connection oriented
applications. We introduce IMS and then look at implementation
agreements, methods and protocols for transport of voice over
MPLS (Multi Protocol Label Switching) networks.
Index Terms- IP multimedia subsystem, VoMPLS, T-MPLS,
Quad Play, QoS, jitter,, A2oMPLS
T has defined T-MPLS as a simplified version of MPLS for
carrier transport networks. T-MPLS and its unique placement
in the telecommunications industry are discussed briefly.
Next, therotl a
as evolved through
ardsfr V oM agreedic ss
carrier implementation agreements
facilitated by IP/MPLS Forum [2].
II. IP MULTIMEDIA SUBSYSTEM
IP Multimedia Subsystem is SIP (Session Initiation
Protocol) and IP based next generation architecture for
unifying the data and media services for mobile and fixed
network users. IMS is being developed by 3GPP, a liaison
body that works with other organizations including TISPAN
(Telecom and Intemet converged Services and protocols for
Advanced Networks).

IMS-S
I. INTRODUCTION
oice over IP has been around for several years. Users
[have experimented with various audio-conferencing tools
offered by messaging services. Discounted phone cards that
route calls through the Internet have been sold for quite some
time. However, the end to end VoIP telephony did not achieve
notable success until recently. The number of worldwide VoIP
customers has now reached 40 million and it is projected to
grow to approximately 250 million by the end of 2011 [1].
The quad play (voice, video, wireless and data) services are
being defined for next generation integrated networks in IMS
(IP Multimedia Subsystem). This tutorial paper discusses the
IMS specifications and focuses on the techniques and methods
developed for transport of voice over MPLS. The protocols
are developed by 3GPP (3rd Generation Partnership Project),
IETF (Internet Engineering Task Force), IP/MPLS Forum
(formerly MFA Forum / MPLS Forum) and the ITU-T
(Telecommunication Standardization Sector of the Figure 1: IMS Architecture
International Telecommunication Union).
IMS goals include provisioning QoS, enabling network
In the next section, the key proposals of IMS are defined. usage billing for media applications and integrating various
MPLS serves as an enabling technology for QoS, and it works media services. IMS adds a control layer below the services
and defines interfaces to the mobile and landline phone
network as well as the IP network. As shown in Figure 1, IMS
Manuscript received November 2008.
Junaid Ahmed Zubairi is with the Department of Computer Science, State defines CSCF (Call Session Control Function) server as the
University of New York, Fredonia, NY 14063 USA ( phone: 716-673- primary SIP server that has three functions. The Proxy CSCF
4694; fax: 716-673-4821; e-mail: Zubairi@fredonia.edu). 2008 IEEE. Personal use of this material is permitted. However,
permission to reprint/republish this material for advertising or
promotional purposes or for creating new collective works for
25 resale or redistribution to servers or lists, or to reuse any
copyrighted component of this work in other works must be
obtained from the IEEE .
lets client user agents connect to the IMS , the Serving CSCF
provides the basic signaling for IMS session and Interrogating
CSCF bridges the different carriers thus enabling roaming.
HSS (Home Subscriber Server), in association with SLF
(Subscription Locator Function) , runs a database listing users
and their subscribed services. HSS would be used to
authenticate the users that request IMS session. IMS links to
all types of networks including wired IP network, wireless
CDMA/GPRS network and fixed phone network. PSTN is
supported by MGCF (Media Gateway Control Function)[3].
IMS uses IETF standardized protocols in order to make it
widely acceptable. However, the details of implementation are
not specified, thus leaving it flexible. Vendors may come up
with products that are IMS-compatible but the same may be
implemented with different strategies. Only the interfaces with
network may follow IMS standards.
III. MPLS AND DIFFSERV PROTOCOLS
Real-time voice traffic has tight timing budget with the
required end to end delay values below 150ms and the jitter or
delay variation around 50ms. The best effort protocols cannot
guarantee such limits because the datagrams do not follow a
fixed path and may arrive at the destination out of order. With
increased congestion, the queues get longer resulting in
increased jitter. These problems make conventional IP
networks largely unsuitable for connection-oriented
applications such as interactive real-time voice. The real-time
audio-conferencing applications written for the conventional
IP networks use forward error correction and inter-mixing of
small packets to recover from packet loss, often leaving the
perceived call quality as less than desirable.
MPLS [4] has emerged as the key integration technology
for carrying voice, video, and data traffic over the same
network. In an MPLS-enabled network, LSPs (Label Switched
Path) are installed from an ingress node to an egress node
prior to start of transmission. Each LSP can be specified with
features that include time constraints and reliability.
Therefore, the connection-oriented applications can take
advantage of the "virtual connections" set by MPLS that
satisfy some constraints. Since the LSPs are stackable, traffic
from different flows sharing some common characteristics can
be aggregated on an LSP. These characteristics may include
common egress and identical QoS and protection
requirements.
MPLS operates by defining a label inside MPLS "shim
header" that is placed on the packet between the layer-2 and
layer-3 headers. The 32-bit shim header is organized as shown
in Figure 2. It is referred to as "shim" header as MPLS allows
la yer-2 and layer-3 to fit together firmly.
MPLS Ladbel (20 TC (3 | (1* TTL (8 |
I bits) bits) bit) bits)
netw
Figure 2: MPLS header fields
26
The MPLS header fields include an unstructured 20-bit
label that identifies the LSP to which this packet belongs. The
label is assigned based on the FEC (forwarding equivalency
class) of the traffic trunk. It is followed by 3 bits TC (Traffic
Class) field and 1 bit indicating whether this is the bottom of
the label stack. The TTL (Time to Live) field has the same
usage as the TTL field in an IP header.
Class based queuing (CBQ) is invoked on routers that
implement the differentiated services (DiffServ) protocol to
create separate queues for each class of traffic. DiffServ
divides the traffic into expedited forwarding (EF) and assured
forwarding (AF) per-hop behaviors (PHBs). EF is the
premium service offered under DiffServ. It is suitable for low
latency and low jitter flows that maintain almost constant rate.
In an MPLS-DiffServ network, the routers jointly implement
various MPLS and DiffServ functions. The ingress router is
responsible for determining an LSP for a new flow request.
The QoS requirements of the new flow can be translated into
DiffServ class assignment at the ingress. For this purpose, the
TC (Traffic Class) field in the MPLS shim header is used. As
the TC field is 3 bits in length, it can represent only eight
different scheduling and drop precedences. Under MPLS-
DiffServ [5], two types of LSPs are defined. E-LSP interprets
the label field of MPLS shim header as the egress identifier
and the TC field as the DiffServ PSC (PHB scheduling class)
combined with the drop precedence. On the other hand, L-
LSP interprets the label field as the DiffServ scheduling
priority and the destination. The TC field in L-LSP is used to
indicate only the drop precedence of the packet. The main
difference between E-LSP and L-LSP is the aggregation
feature in E-LSP resulting in scalability; however, some PSCs
in an E-LSP may suffer because the bandwidth is reserved for
the whole LSP [6]. Figure 3 shows the placement of the
DiffServ class identifier and MPLS label for an E-LSP.
MPLS LAbel (20 TC S TTL
bits) (FEC (DiffSCrv (1 bit) (8 bits)
Dti))
IV. TMPLS PROTOCOL
ITU-T believes that the transport networks do not have
highly varying traffic patterns thus there is no need for some
of the features of MPLS that largely suit the dynamically
changing traffic of Internet. They introduced T-MPLS in 2006
for support of connection oriented packet switching flows.
The T-MPLS (Transport MPLS) (ITU-T G.8110.1) protocol
implements a simplified MPLS architecture for carrier class
networks. T-MPLS adds the TE capabilities of SONET/SDH
to Layer 2 network designs [7]. It provides backup tunnels
with 50ms protection switching, thus enabling the migration
from SONET/SDH transport to a fully packet switched
ork ith com parable reliability[8].
The standard documents approved were G.81 10.1
Architecture of T-MPLS layer network, G.8112 Interfaces for
the T-MPLS hierarchy (TMH) and G.8121 Characteristics of
T-MPLS equipment. T-MPLS is described as a connection
oriented packet transport technology that uses the MPLS label
swapping and forwarding, carrier networks performance
monitoring, simplified protection switching and
ASON/GMPLS control and management functions [9]. It
involves setting up unidirectional and bidirectional P2P (Point
to Point) connections that take identical network path in both
directions and receive TE support and management. T-MPLS
simplifies end to end OAM (Operation, Administration and
Management) by removing the IP specific functionality. For
example, T-MPLS removes the PHP (Penultimate Hop
Popping) feature that would require IP processing of the
packet at the end router and similarly the ECMP feature is
removed in order to avoid possible source identification
confusion. The T-MPLS tunnel can carry multiple L2 and L3
services including IP/MPLS LSPs and PWE3 pseudowires.
Companies that deploy T-MPLS are expected to find it easier
and economical in terms of man-hours of staff training, as
compared with IP/MPLS networks. It is because of the fact
that the T-MPLS network's architecture is similar to
SONET/SDH network.
Recently, in early 2008, ITU-T and IETF have deliberated
to resolve inconsistencies between MPLS and T-MPLS and
the consensus is emerging to define separate code points for
MPLS and T-MPLS in order to avoid confusion. IETF and
ITU-T agree that MPLS and T-MPLS are disjoint networks.
An LSP initiated from either network would encapsulate into
Ethernet before transiting the other network. Client support in
T-MPLS is based on IETF Pseudo wire model for Layer-2
VPN. The two-layer architecture includes top layer as client
Virtual Circuit LSPs and bottom layer for aggregating the VC
LSPs into Trunk LSPs. The key differences between MPLS
and T-MPLS include the use of bidirectional LSPs in T-MPLS
and no PHP, no merging of LSPs and no ECMP routing in T-
MPLS [10].
V. VOICE TRANSPORT OVER MPLS
Voice is carried on MPLS network either directly or with
VoIP stack i.e. RTP/UDP/IP/MPLS. On the other hand,
VoMPLS [11,12] carries voice directly over MPLS, reducing
the headers significantly. Figure 4 shows the comparison of
protocol stacks in different schemes. VoIP has the clear
advantage of providing end-to-end connectivity with mature
protocols. The MPLS user-to-network interface (UNI)
definition for bringing users in direct contact with MPLS
networks has been defined recently [13,14,15]. Using MFA
IA 1.0 and IA 5.0, voice can be directly encapsulated in
MPLS packets. The other option is to use the ATM over
MPLS definition to enable VoATM voice service. Among
these choices, VoMPLS is the most promising solution as it
involves the least amount of protocol tax.
A. VoMPLS
In VoMPLS,-ktd the
lt voice
r packets are directly transpo ed over
27
the MPLS bypassing the RTP/UDP/IP protocol stack. As seen
in Figure 4, the VoMPLS protocol stack is more compact as
compared to VoIP. The MPLS UNI (User Network Interface)
definition [14] allows users at CE (Customer edge) to
establish LSPs to the PE (Provider Edge) network. The LSP
may be a single link or a layer-2 network but it is considered a
single hop LSP. Since there may be thousands of CE-to-CE
connections, a potential scalability issue arises. The MPLS
proxy admission control [15] solves the scalability problem by
letting the PE ingress router search for existing tunnels that
can satisfy the traffic parameters TLV in the label request
message sent by the CE. Once a suitable LSP is found, its RIL
(Resource Index Label) is returned to the CE which then uses
it to encapsulate the traffic for that tunnel. QoS is guaranteed
only within the PE-to-PE network. For CE-to-PE links,
customers have to deploy layer-2 QoS techniques for
provisioning of resources.
o MA JA M ALIA AL3
RT1PLS MPL5 A
IJDP
Figure 4: VolPoMPLS and VoMPLS protocol stacks
B. MPLS UNI With Proxy Admission Control
In voice deployment over MPLS, the VGW (Voice
Gateway) acts as the CE requesting guaranteed LSPs to the
remote VGW. The MPLS proxy admission control allows the
voice gateways to dynamically request LSPs to remote VGWs
and to share the multiplexed LSP-TE tunnels among
themselves through the local PE acting as a proxy for
admission control [16]. Using voice gateway as an example as
in Figure 5, the following exchange of control messages may
occur between the VGW and PE before a call can be
established:
1) Phone 1 requests a voice call to Phone2.
2) VGW sends a label request message to PE containing
FEC (Forwarding Equivalence Class) and Traffic Parameters
TLV for the intended call.
3) PE acts as a proxy for the local VGW and searches
existing TE tunnels to remote VGW that satisfy the traffic
parameters and FEC in the label request message
4) If an existing tunnel is found that can satisfy the request,
PE sends its RIL to the requesting VGW
5) The requesting VGW can now initiate the call and then
release the resources once the call is terminated.
From the above, it is obvious that the MPLS UNI together
with MPLS Proxy admission control provides a powerful
mechanism to deploy thousands of voice calls through the PE
network without control messaging overflow.
 C. VoMPLS Header Formats
In VoMPLS, Various header formats are defined for
different payload types. Payloads may include encoded audio
information, dialed digits, silence descriptors and control
signaling. Two types of frames are defined in VoMPLS IA
1.0: primary and control frames. The VoMPLS primary frame
header is shown [10] in Figure 6. Many primary VoMPLS
frames may be multiplexed within a single MPLS packet.
Pb-
Pbetween dialed digits and the channel related signaling. Time
Figure 5: MPLS UNI and Proxy Admission Control
Channeil Pay Counte Payload Pad
Wfidtifie
(8 bits)
load
Type
(8 bits) Length
(6 bits)
Length
(2 bits)
bits)
Figure 6: VoMPLS MFA 1 primary frame header fields
VoMPLS primary frame header is 4 bytes long and includes
5 fields:
> Channel Identifier uniquely identifies the voice
channel that is the source of the payload. Thus, a
total of 248 different voice calls can be multiplexed
into a single LSP.
Payload Type identifies the encoding scheme used as
well as silence removal/insertion descriptors. A value
equal to or above 224 indicates control payload (part
of the control frame) that would allow DTMF (dual
tone multi frequency) dialed digits as well as
signaling for the channel to be carried [4].
> The revolving Counter field is set at the first sample
or frame and keeps incrementing for each additional
frame.
> Payload Length is read in conjunction with the pad
length to keep the payload a multiple of 4 bytes.
Multiple primary frames may be multiplexed with the use of
optional inner MPLS labels in addition to one mandatory
outer MPLS label. Using CID, up to 248 voice calls can be
aggregated. The payload of primary frame may consist of
encoded audio data or SID (Silence Insertion Descriptor)
parameters.
Keeping the frames and header formats in perspective, the
voice calls can be established as follows [4].
28
1) Using MPLS signaling mechanism, a bidirectional LSP is
created
2) As voice call request arrives at the edge of the MPLS
network, an existing CID is allocated to the new call within
the LSP just created or CID signaling is done to establish the
new channel.
3) Optionally, inner LSPs may be created within the outer LSP
in which case the outer LSP label, inner LSP label and CID
unqul idntf th voc. al
uniquely identify the voice call.
The VoMPLS control frame header is also 4 bytes in
length. Control fames cannot be multiplexed and must be
carried separately however the mandatory outer label and the
optional inner label precede the control frame header fields in
order to uniquely identify the voice call for which this control
frame is sent.
In the control frame, the Payload Type field distinguishes
Stamp is relative to the first randomized time stamp. The
Redundancy field is very important to ensure the receipt of the
control frames. If the Redundancy field is set to 0, 1, or 2, the
control packet is repeated that many times.
Another solution for VoMPLS reuses ATM Adaptation
Layer 2 (AAL2) components but replaces ATM by MPLS
[6,1 1], thus eliminating the ATM cell overhead. The MFA 5.0
implementation agreement proposes voice trunking over
MPLS by directly encapsulating AAL2 common part sublayer
(CPS) packets into MPLS (A2oMPLS). The gateway to the
MPLS network should be able to function as an AAL2 switch.
Multiple A2oMPLS connections may be multiplexed into a
single LSP. One MPLS frame may carry multiple CPS
packets. The A2oMPLS sub-frames may be of different length
but the maximum CPS packet payload length is restricted to
between 45 octets to 64 octets. Like the MFA IAlO, the CID
field in the CPS packet header allows up to 248 A2oMPLS
connections to be multiplexed. Each A2oMPLS connection
can be uniquely identified with outer label of the LSP,
optional inner label and the CID value. When inner labels are
used the number of calls that are carried by a single LSP
.
increases rapldly.
r
The following fields are included in the A2oMPLS header:
Reserved (10 bits) currently ignored.
L
> Length (6 bits) used for padding length. It is set to
0 if A2oMPLS packet length exceeds 64 bytes.
> Sequence Number (S No) (16 bits) used if
guaranteed ordered packet delivery is required.
Sequence number of 0 indicates otherwise.
The following fields are inside the CPS packet header;
several CPS packets may be packed in one MPLS frame.
> CID (8 bits) identifies the A2oMPLS connection
carried. Thus a total of 248 connections (8 to 255)
can be multiplexed into a single LSP.
> LI (6 bits) identifies the length of the CPS packet.
> UUI (5 bits) is used for user-to-user indication (0
to 27 for users, 30-31 for layer management, and
bv 1rsre)
 HEC (5 bits) (header error control) uses CRC
checksum. However, this field may not be used in
cheksu.'Hweer,thi feldmaynobeusei
the MPLS environment.
VI. CONCLUSION
Industry is moving towards integration of phone system
with the packet switching network. IMS is being developed to
provide uniform interface for users from cellular, landline or
wired systems. In wired and wireless packet based networks,
MPLS can work with other protocols to guarantee the
bandwidth needed for a voice call and provide fast protection
switching. Voice can be deployed over MPLS using a variety
of techniques that have been developed recently. In this paper,
these techniques including VoIPoMPLS and VoMPLS are
discussed.
VoIPoMPLS deploys voice over the RTP/UDP/IP protocol
stack, which uses MPLS tunnels. Since the number of
protocols involved iS large, the control information equals or
exceeds the payload. Most efficient method is to run voice
directly over MPLS. VoMPLS is discussed in detail including
various proposals for enabling it in the network. VoMPLS is
well suited for multiplexing in the core and carrying the bulk
of voice calls through the MPLS domain. The difference
between MPLS and T-MPLS, a new ITU-T proposal for
carrier networks, is highlighted.
The MPLS UNI proposal presented in 2004 is outlined.
Since the original proposal was not scalable, proxy admission
control was introduced to pack new control existing
LSPs intointrodu d
tunnels.
With the introduction of IMS, many technologies would be
integrated to allow exciting new features in packet voice. For
example, transparent connectivity through landline and mobile
networks would let the user continue the conversation while
changing the connection from landline network to mobile
network or vice versa. Since IMS is based on IPV6, the IMS
adoption would be slow in parts of the world where IPV4 still
runs the network. The VoP phones would be flexible and
allow all the functionalities of current PSTN phones. Security
of packet based phones and authentication of users poses new
challenges that must be analyzed and resolved for widespread
deployment.
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Junaid Ahmed Zubairi received his BE (Electrical Engineering) from NED
University of Engineering, Pakistan and MS and Ph.D. (Computer
Engineering) from Syracuse University, USA. He worked in Sir Syed
University Pakistan and Intl' Islamic University Malaysia before joining
SUNY at Fredonia in 1999 where currently he is an Associate Professor in the
Department of Computer Science. Dr. Zubairi is a recipient of many awards
including Malaysian Government IRPA research award, National Science
Foundation MACS award, SUNY Scholarly Incentive award and SUNY
Faculty Fellowship award. He has published several book chapters, journal
articles and conference papers in his areas of interest. His research interests
include network applications, traffic engineering and performance evaluation
to
of networks. He can be reached at junaid.zubairi fredonia.edu.

accessed Nov 4 2008

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