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10
Evolution of the
voice interconnect
The migration to all-IP networks calls for an evolution of the interconnect between operator
networks. This might lead to a new interconnect architecture featuring an intermediate
carrier network based on either a pure IP transit architecture or a service hub architecture.
has resulted in a reduction of the international interconnect price of around
7 percent per year.
J oh n Ba l dw i n, Jörg E w e rt a n d S on e r Ya m e n
During the past decade, network
operators have transformed their
existing wireline and wireless
core networks from Time Division
Multiplex (TDM) or Asynchronous
Transfer Mode (ATM) transport
technology to IP transport.
The evolution of the voice
interconnect between operator
networks to IP is another natural
step towards an all-IP archi­
tecture. This transition will result
in more efficient bandwidth
usage, better speech quality,
reduced costs, and a clear migration path to a full multimedia
service interconnection.
Background
Traffic forecast
During the past five years there has
been a continuous increase of 10-15
percent per year in international interconnect traffic, and mobile operators
BOX A are playing a more important role in
interconnect for two main reasons.
Firstly, worldwide the number of
subscribers in mobile networks is
still increasing, mainly driven by the
demand in developing and emerging
markets with low fixed-line penetration. Secondly, there is an increased
trend towards fixed-mobile substitution in developed countries.
In 2002, the number of mobile
subscribers worldwide surpassed the
number of fixed subscribers worldwide.
The increases in subscribers and minutes of use have resulted in the mobile
terminating traffic surpassing the fixed
terminating traffic in 2009.
Although traffic growth continues,
PC-based voice clients such as Skype
have started to use the internet to
bypass traditional international transit networks. These new entrants have
managed to capture approximately
8 percent of international voice traffic
volume.1 The increased competition
Existing interconnect architecture
Today’s interconnect architecture is
predominantly TDM-based and is a mixture of direct interconnects between
operator networks, typically used within a country, and indirect interconnects
that use an intermediate global carrier
network to reach the rest of the world.
With the migration towards an IP-based
interconnect, operators and carriers
are looking at new architectures rather
than simply replacing TDM pipes with
IP pipes (Figure 1).
The transformation from existing
ISDN User Part (ISUP)/TDM interconnects to an IP-based interconnect has
the following advantages:
Evolution
of the voice interconnect:
Using bandwidth-optimized codecs will
reduce bandwidth needs, and using
IP instead of TDM will ensure better
utilization of the point of interconnect.
Both will significantly reduce capital
Terms and abbreviations
AMR
AMR-NB
AMR-WB
ATM
BICC
CELP
CPC
CS-IBCF
EFR
GRX
IBCF
IMS
IPX
Adaptive Multi-Rate
Adaptive Multi-Rate Narrowband
Adaptive Multi-Rate Wideband
Asynchronous Transfer Mode
Bearer Independent Call Control
code-excited linear prediction
Continuous Packet Connectivity
Circuit Switched Interconnection
Border Control Function
Enhanced Full Rate
GPRS Roaming Exchange Node
Interconnection Border
Control Function
IP Multimedia Subsystem
Internet Packet Exchange
E r i c s s o n r e v i e w • 2 2010
ISUP
MGw
M-MGw
MMTel
MMS
MNO
MSRP
MSC-S
MSS
MTP3
NNI
PCM
PLMN
RANAP
ISDN User Part
Media Gateway
Mobile Media Gateway
Multimedia Telephony
Multimedia Messaging Service
mobile network operator
Message Session Relay Protocol
Mobile Switching Center Server
mobile softswitch
Message Transfer Part 3
Network to Network Interface
pulse-code modulation
Public Land Mobile Network
Radio Access Network
Application Part
RCS
RTP
RTCP
SCCP
SCTP
SIP
SIP-I
TDM
TrFO
TrGw
TCP
UDP
VoIP
Rich Communication Suite
Real-time Transport Protocol
Real-time Transport
Control Protocol
Signaling Connection Control Part
Stream Control Transmission Protocol
Session Initiation Protocol
SIP with encapsulated ISUP
Time Division Multiplex / Multiplexing
Transcoder Free Operation
Trunking Gateway
Transmission Control Protocol
User Datagram Protocol
Voice over Internet Protocol
11
expenditure (capex) and operational
expenditure (opex).
Interconnect architecture:
IP technology offers higher flexibility
in choosing interconnect partners
and architecture.
Multimedia interconnect: Operators
have started launching new multimedia
based services such as presence and
messaging. One of the key success
factors will be interoperability between
operators. This will require evolving the
interconnect from a pure voice interconnect to a combined voice and multimedia interconnect.
Figure 1 Evolution of the interconnect from TDM to IP is the natural next step after the
transition to IP within operator domains.
Mobile
softswitch
MSC-S
Control
TDM
IP
Softswitch
or IMS
IP
M-MGw
TDM Gw
MSC-S
Control
The following sections examine the
three areas in more detail.
Evolution of the voice interconnect
Today, voice interconnect is handled
using legacy TDM connections with
ISUP signaling and the G.711 codec. Over
the years ISUP signaling has matured
and now supports national and international variants, as well as redundancy mechanisms required for telecom
grade service.
However, when it comes to the introduction of new services and inter­
working with IP-based networks, ISUP
has severe limitations and adds to the
complexity of the solution.
Figure 2 shows a Session Initiation
Protocol with encapsulated ISDN User
Part (SIP-I) interconnect between two
MSS domains. When SIP-I is used, in
addition to the protocol mapping done
for SIP, the complete ISDN User Part
(ISUP) message is transferred within
the body of the SIP message. This allows
the receiving SIP User Agent to process the encapsulated ISUP messages,
so that no service-related information
is lost for legacy functionality.2 Similar
to ISUP, market variants are supported using SIP-I screening, which allows
modification of messages, parameters
and parameter fields from the encapsulated ISUP content.
Figure 3 shows an SIP inter­connect
between an IMS and MSS domain.
When the Gateway carries out protocol
interworking between SIP and ISUP or
Bearer Independent Call Control (BICC)
or Radio Access Network Application
Part (RANAP), it performs a mapping
between the message parameters of
the SIP and ISUP protocol.3 This mapping can result in information loss
Mobile
softswitch
IP
IP
IP
IP Gw
M-MGw
Figure 2 (NNI).
Softswitch
or IMS
Reference architecture of an SIP-I and SIP Network to Network Interface
SIP-I Network to Network Interface
SIP-I
UDP/TCP/SCTP
IP
Ethernet
Nc
Nc
MSC
server
BICC
H.248
Mc
MSC
server
H.248
Mc/Mn
CS
IBCF
CS-Ix
CS
IBCF
CS-Ix
MSC
server
H.248
Mc/Mn
Nb
MGw
BICC
MSC
server
H.248
Mc
Nb
MGw
CS
TrGw
CS
TrG
RTP/RTCP
UDP
IP
Ethernet
MGw
MGw
Control plane
User plane
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12
Figure 3 (NNI).
Reference architecture of an SIP-I and SIP Network to Network Interface
SIP Network to Network Interface
SIP
UDP/TCP/SCTP
IP
Ethernet
Nc
MSC
server
CS
IBCF
IMS or
VoIP networks
CS-Ix
BICC
MSC
server
H.248
Mc
H.248
Mc/Mn
Nb
CS
TrGw
MGw
RTP/RTCP
UDP
IP
Ethernet
MGw
Control plane
User plane
Figure 4 IP technology will be used as indirect, direct and indirect service aware
interconnection.
MNO
FNO
SIP
Hub
MNO
Global carrier
IP networks
SIP
Hub
FNO
MNO
FNO
MNO
E r i c s s o n r e v i e w • 2 2010
Direct IP interconnection
Indirect IP interconnection via IP transport service
Indirect IP interconnection via Hub service
Fixed network operator
Mobile network operator
related to the requested service and
does not support all legacy services.
For voice interconnect between networks there is more to consider than
just the choice of signaling protocol.
The following aspects should also be
considered for interconnects based on
SIP-I and SIP:
ignaling policing: policing is needed to
S
protect the domain against unwanted
messages. For ISUP interconnects,
policing is performed on the MTP3 or
SCCP protocol level. For an SIP or SIP-I it
is done by the Interconnection Border
Control Function (IBCF) on the
SIP/SIP-I signaling level and by the
Trunking Gateway (TrGw) on the media
protocol level.
Bandwidth management: in the case of
an ISUP interconnect, there is always a
predefined bandwidth usage of 64Kbps
or 56Kbps for G.711 per channel. Due to
the fixed timeslots on the payload path it
is not possible to use more than the
assigned bandwidth. For an IP-based
interconnect the bandwidth needed
depends on the codec used. Also the
user plane connections share a common
IP pipe. This means that proper bandwidth allocations and admission control
mechanisms are needed to avoid exceeding the allocated bandwidth or causing
deterioration of the voice services.
Voice transcoding: as more codecs are
now available, it is important to introduce negotiation mechanisms to select
the best codec for each situation. This
will reduce the number of transcoding
stages, resulting in a Transcoder Free
Operation (TrFO) whenever possible.
Fax, Video and CS Data Calls: fax and
data calls represent only a small fraction
of the total number of calls but must still
work properly. In order to support compressed speech without disturbing fax
and modem operation, voiceband data
traffic detection and fallback to G.711 is
desired. Also the T.38 fax gateway is
desired to support proper fax inter­
working over poor quality IP transport
connections.
Charging and Accounting: SIP/SIP-I
based interconnect will reuse the existing charging mechanism based on call
data records.
Emergency and Priority calls: interconnects need to support emergency and
priority calls. Emergency and priority
calls are indicated with a certain ISUP
13
parameter (Continous Packet
Connectivity or CPC) and a certain SIP
parameter.
Operators may implement IP-based
voice interconnect using different codec
types (Table 1) and enjoy the benefits
of each. These can also be seen as evolution steps.
Usage of IP transmission (still G.711)
If transmission efficiency is not so
important, an operator may introduce
IP-based interconnect without support for any additional codec above the
default G.711 codec.
sage of bandwidth efficient codecs: for
U
example, G.729 with transcoding
As a next step, codecs with better bandwidth characteristics can be introduced
on the interconnect to reduce opex. For
example, G.729 is the most commonly
used fixed network codec and can be
used for compression.
emoval of the need for transcoding
R
(end-to-end codec negotiation)
If the same codec is supported within
the two connected networks then it is
desirable to support this codec also on
the interconnect. This can eliminate the
need for transcoding, or at least minimize the number of transcoding stages,
ensuring resource efficiency in MGws,
which reduces capex and improves
speech quality. As an example, AMR
coded speech, which is commonly used
by mobile networks, can be used on
interconnect to achieve TrFO end-to-end
when connecting two mobile networks.
TrFO reduces the need for transcoders
and therefore reduces capex.
I mproved voice quality with High
Definition (HD) voice (for example, AMRWB, G.722)
Once interconnect is on IP transport
and TrFO is achieved, new codecs with
superior speech quality, HD-voice, can
be introduced also for inter-Public Land
Mobile Network (PLMN) calls. AMR-WB,
which is optimized for mobile access,
and G.722, which is widely used for fixed
access, can be used for superior speech
quality. These codecs can be transcoded to each other without losing wideband characteristics. HD voice on interconnect enables superior speech quality for calls between mobile and fixed
access and can help to increase revenue
and reduce churn.
Table 2 summarizes which codecs
Table 1: Most popular codecs and their characteristics.
CODEC
EXPLANATION
PACKETIZATION
TIME /DEFAULT
VALUES IN MS)
EFFECTIVE
DATA RATE
(KBPS)
G.711
G.711 is the default codec for fixed and mobile applications and uses pulse-code
modulation (PCM). It provides the best speech quality for narrowband codecs
with a user data rate of 64K.
0.125
64 (over TDM)
G.729
G.729 is one of the most popular narrowband codecs for IP-based fixed networks
based on code-excited linear prediction (CELP). It provides similar speech
quality characteristics to AMR 7.4 with a user data rate of 8K.
20
23 (over IP)
G.722.0
G.722.0 is the fixed rate 7kHz wideband audio codec with a user data rate of 64K.
Due to extended audio spectrum, it provides excellent audio quality with volume
and clarity.
20
54 (over IP)
AMR-NB
Adaptive multi-rate (AMR) is the most popular narrowband codec for wireless
networks. It adapts to radio conditions and capacity requirements with user data
rates between 12.2K down to 4.75K.
20
23 (over IP)
AMR-WB
Adaptive multi-rate wideband (AMR-WB) is the 7kHz wideband audio codec with
similar speech quality to G.722.0. It adapts to radio conditions and capacity
requirements with user data rates between 12.65K down to 6.60K.
20
23 (over IP)
Table 2: Summary of the codecs supported per domain.
MOBILE SOFTSWITCH
Inside MSS
G.711
WIRELINE SOFTSWITCH
IMS
G.711
G.711
G.711
G.729
G.729
Towards other networks
G.729
AMR-NB
AMR-NB
AMR-NB
AMR-WB
AMR-WB
AMR-WB
GSM EFR
GSM EFR
are used in which domain. It is clear
that G.711 is the only codec that is
common for all domains. A common
codec via end-to-end codec negotiation
is possible with AMR-NB, AMR-WB for
a MSS-MSS and MSS-IMS interconnect.
This is not possible for an MSS-wireline
softswitch interconnect; G.729 is the
optimal choice in this case, but requires
transcoding in MSS due to the unavailability of the G.729 codec in mobile
phones.
Interconnect architecture
With the migration to all-IP networks
and IP-based interconnect, operators
and carriers are looking at new interconnect architectures, rather than simply replacing TDM pipes with IP pipes.
In the new architectures, the concept of
an intermediate carrier network exists
in two forms:
ure IP transit architecture: in this case,
P
the carrier network essentially provides
an IP interconnection “pipe” between
operators, which will have a bilateral
agreement for settlement. The carrier
network is agnostic to the type of
application or media flowing through the
network and will typically charge the
operators based upon the amount of
data transferred.
Service Hub architecture: in this case,
the carrier network is service aware and
can differentiate the type of application
or media that is flowing through the
network for QoS and charging purposes.
The carrier network can provide a
complete multilateral hubbing service
where operators set up single contracts
with carriers for their complete inter­
connection needs.
The future IP interconnect architecture is likely to involve a combination of:
Direct
IP interconnect between two
mobile operators (National traffic)
Indirect interconnect via a third-party
carrier IP backbone (International traffic
– first using pure IP transit)
Indirect interconnect via a third-party
carrier network that also is service
aware (Service Hub)
In its pure IP transit form, as illus­trated
in Figure 4, the carrier network
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14
Figure 5 Combined voice and multimedia interconnect.
MSS
IMS
Control plane
CS SIP-I voice/video
IMS SIP instant messaging
IMS SIP presence
IMS SIP voice/video
Multimedia IP
interconnect
MSS
IMS
User plane
MSRP
RTP
provides a mobile/fixed network
operator (MNO, FNO) with IP connectivity to all other operators around the
world, eliminating the need to establish
individual IP connections. The IP
connection provided by the carrier has QoS characteristics defined, so
can be used for real-time services such
as voice.
In its service hub form, the carrier
network can provide additional services such as service-based inter-operator
settlement, service based routing and
service interworking. This allows fixed/
mobile operators to delegate a large part
of interconnect administration to their
chosen carrier.
The different architectures give network operators flexibility in choosing
the appropriate interconnect archi­
tecture for particular destinations
according to traffic and commercial
requirements. The usage of IP technology gives higher flexibility, such as
having fewer and larger points of interconnection. From an economy of scale
perspective, this can provide benefits
in terms of reduced administration and
interconnect fees.
These architectures provide welldefined technical and business roles
for mobile/fixed operators and also for
the global carriers. This is necessary for
a sustainable interconnect solution. It
is unlikely that models based upon
interconnections over the public interE r i c s s o n r e v i e w • 2 2010
net can provide such a sustainable interconnect solution.
There is a need for industry-wide coordination as operator network IP-based
interconnects are emerging. The GSM
Association is active in coordinating the
industry’s thinking on IP-based interconnection. The GSMA suggests that the
existing carrier GRX IP networks are
evolved to support voice and multimedia services. The IPX project builds upon
the GRX to add QoS and so supports realtime CS and IMS services such as voice,
video and presence. It is expected that
within IPX there will be a service hub
role for CS and IMS services. The service
hub role is beneficial because an operator only has to establish one agreement
with the IPX hub provider who then has
agreements with all other operators. To
provide a service hub function for these
real-time services SIP Servers will be
introduced into the IPX core
Multimedia interconnect
Once IP-based interconnect for voice
is established, there is a natural evolution from voice to multimedia across
interconnects, shown in Figure 5. The
Rich Communication Suite (RCS) initiative from GSMA provides enrichment
to voice services. RCS also provides enduser services such as presence, instant
messaging and video sharing.
The value of the RCS services will
increase with the number of users and
also with the number of interconnected networks (Metcalfe’s law). Operators
benefited from this correlation in the
past, when they started their SMS and
MSS services, and at the same time
secured interoperability. This requires
RCS services to be supported across operator interconnects. These interconnects
could be direct between two operators,
or indirect using an intermediate carrier IP network, such as IPX.
RCS is a set of multimedia services based upon 3GPP IMS using SIP
signaling and RTP and MSRP media.
The interconnect needs to support these
signaling and media protocols. RCS is
likely to be the first service that makes
IP interconnect mandatory, simply
because TDM networks cannot transport the media types defined in RCS.
The different types of media in RCS
have different bandwidth and delay
requirements – video requires high
bandwidth for media, presence has a
high signaling rate. It is not sufficient to base inter-operator settlement upon packet count at IP level.
These services require new settlement
arrangements between operators
and new KPIs to be defined for interconnects. There could be different
settlement policies for different RCS
services. The session border gateway has an important role to play in
policing KPIs.
In effect, operator interconnects and
intermediate carrier networks need
to be service aware, both in terms of
technical realization and commercial
settlement mechanisms. This will allow
RCS services to be deployed at national and international level, and will
be the basis for the evolution of these
networks and interconnects towards
the full MMTel service.
With the introduction of new multi­
media services there are benefits to be
gained from using the service-aware
hub carrier network. The service hub
can provide interworking between
services both at the media layer (trans­
coding) and also at the service layer – for
example, interworking between legacy
SMS/MMS and IMS messaging.
Conclusion
The evolution of the voice interconnect
creates opportunities for the operators to:
reduce capex/opex by using IP technolo-
15
gy with more efficient codecs
optimize voice quality by reducing the
transcoding steps and introducing new
HD audio codecs
reduce transmission costs by selecting a
new interconnect architecture
increase revenue by providing multi­
media interworking between operator
networks.
Ericsson believes that the evolution
of the voice interconnect is an important part of the overall evolution of tele­
phony. Ericsson is working with the
major operators, carriers and standards
bodies to ensure that this evolution is
successful and that end users and the
industry can benefit from the opportunities described above.
John Baldwin
Jörg Ewert
joined Ericsson in 2006
following the Marconi
acquisition. He first
worked for the Systems
and Technology group within the Core
& IMS unit of Business Unit Networks
on strategies for connecting the
mobile softswitch (MSS) to IMS
networks. He was then involved in IPbased interconnects for MSS and IMS
networks. At present, he is involved in
the study of IP-based interconnects
and new applications that use IMS
networks and the internet. He holds
a Bachelor of Science in Mathematics
from the University of Warwick, UK.
joined Ericsson in 1999
as a system manager for
MSS operation and
maintenance. He later
became the leader of the O&M system
design team. In 2005 he joined the
MSS product management team,
where he was responsible for network
management and security. At present,
he is responsible for MSS and IMS
co-existence and SIP interconnects in
product line switching at Business Unit
Networks. Jörg has studied business
administration at the University of
Hagen, Germany, and he holds a
Master of Science and a PhD. in
Physics from the University of
Göttingen, Germany.
Soner Yamen
joined Ericsson in 1995 as
a support engineer. In
1999, after having served
as a solution and project
manager in the market unit, he
transferred to system management at
Ericsson Eurolab. In 2000, he joined
product management, where he is
currently responsible for the functional
areas Transport and Network Quality
& Efficient Infrastructure in product line
switching at Business Unit Networks. He
holds a Master of Science in Electrical
Engineering from the University of
Wisconsin-Madison, US.
References
1. http://www.telegeography.com/product-info/tg/index.php
2.ITU Q1912.5, 3GPP TS 29.164
3. 3GPP TS 29.163
E r i c s s o n r e v i e w • 2 2010