Requirements
and Service Scenario for High Quality
Mobile VoIP Service in Convergence Network
Kyu Ouk Lee, Young Sun Kim
Electronics and Telecommunications Research Institute (ETRI)
Address: 161 Gajeong-dong, Yuseong-gu, Daejeon, 305-700, KOREA
E-mail: kolee@etri.re.kr, Tel: 82-42-860-5756
Abstract - Current IP based voice service has limitation of
QoS owing to the packet delay, loss, and jitter. To improve voice
service quality, along with providing voice mobility functions
over current IP based network, service scenarios are proposed
by providing seamless mobility in heterogeneous network among
WiFi, 3G, 4G, mobile WiMAX. First of all, requirements for high
quality mobile VoIP are mentioned in user terminal layer,
network layer, service control layer, and detail service scenarios
are described in registration phase, service signal phase, real
service provisioning phase, and service closing phase. Currently
VoIP service with WiBro and WiFi is under consideration, so
proposed service scenarios will became useful guidelines for
high quality mobile VoIP service in heterogeneous network.
Keywords - VoIP, Mobility, WiBro, WiFi, 3G, 4G
1. Introduction
Nowadays, infra network structure for voice service has
been transferred from circuit base to packet base and from
fixed base to mobile base. Legacy circuit based voice service
has limitation of service coverage owing to the decrease of
network resource utilizations, while current IP based voice
service has limitation of QoS owing to the packet delay, loss,
and jitter. To improve voice service quality over these kinds
of circuit based and fixed IP based network, service scenarios
are proposed for high quality mobile VoIP services by
providing seamless mobility in convergence network among
WiFi, 3G, 4G, WiBro (Korean version of mobile WiMAX).
Until now on, the main services of WiBro and WiFi are
Internet access services, especially mobility support in case
of WiBro is up to 60km/hour, while main services of 3G and
4G is circuit based voice service and packet based data and
video service. Currently VoIP service with WiBro and WiFi is
under consideration, so this paper will became useful
guidelines for high quality mobile VoIP service in
convergence network among WiFi, 3G, 4G, WiBro
2. Requirements in three layers for high quality
mobile VoIP service
Some definitions and necessary requirements for providing
the high quality mobile VoIP service in user terminal layer,
network layer, and service control layer are as follows.
2.1 Definitions
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* Common Driver: A driver to support multi-band
multi-mode operations when User Terminal moves to
different network by reconfirming and requesting the
identity of User Terminal identities through access
network.
* New IP based mobility function: A new concept
function to support seamless mobility of User
Terminal by using "Make before break" technology in
heterogeneous network. It fix the packet delay, packet
loss, and network disconnection which can be
occurred at current using network structure by using
mobility function as "Mobile IP", "Simple IP",
"Mobile SIP", ''UMA"
* Heterogeneous Network: It can be defined with
several meanings. In this contribution, it is defined as
composed network with WiFi, 3G, 4G, and mobile
WiMAX subscribes and their concerning equipments
* Mobility Control Platform: A platform to update and
manage the mobility information of User Terminal in
heterogeneous network. Mobility Control Platform
use "New IP based mobility function" to support
seamless mobility of User Terminal.
* Network Control Platform: A platform to support QoS
assurance of VoIP and seamless service of User
Terminal in heterogeneous network. This platform
will be developed by enhancing the RACF of FRA
release 1/2 documents (ITU-T, Y.2012).
* Service Control Platform: A platform to support
seamless VoIP service by interworking or exchange
the service control signal and service profile
information between networks. This platform will be
developed by enhancing the "service control
functions" of FRA release 1/2 documents (ITU-T,
Y.2012).
* User Terminal: An equipment that user use for call
connection, conversation, and release as mobile phone,
PDA (Packet Data Assembly), notebook, etc. User
Terminal is composed of Originating User Terminal
and Terminating User Terminal.
2.2 Requirements in user terminal layer
First of all, user terminal should be equipped with wideband
voice codec such as G722.2, G.729.1, along with concerning
network chip as WiBro, WiFi, HSPDA for service providing.
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And, driver platform is also required to support multi-band
multi-mode operation for reconfirmation of user terminal
identities when user terminal moves to different network.
This platform updates and manages the mobile and location
information of user terminal, and provides the QoS of
seamless service by using new IP based mobility function.
2.3 Requirements in network layer
2.3.3 Requirements of network control platform in
network layer
Currently, mobile IP, simple IP, mobile SIP, and UMA
(Unlicensed Mobile Access) technologies are using for
mobility of user terminal. But, these kinds of technologies are
not sufficient to support high quality mobile VoIP service in
convergence network. For example, mobile IP is method of
IP based mobility support by reallocation of different IP
whenever user terminal moves to different network. Normally,
network node as HA (Home Agent), and FA (Foreign Agent)
should be added when mobile IP is used, and packet loss and
packet delay will be occurred owing to the data transmission
through HA, so mobile IP is not suitable for high quality
mobile VoIP service providing. Simple IP is used only inside
same network, so when user terminal moves other network,
current service will be disconnected and is not suitable in
convergence network. Mobile SIP is method of using SIP of
service control signal, so when user terminal moves other
network, current session will be disconnected, and new
session is established by new IP address allocation. Packet
delay and loss will be occurred owing to the session
reallocation in case mobile SIP is used, so mobile SIP is also
not suitable in convergence network. While, UMA
(Unlicensed Mobile Access) is mobility support method by
using UNC (Universal Naming Convention) server for
seamless interworking between mobile network and WLAN
(Wireless Local Area Network) which has not control signal.
UMA is not suitable for interworking between mobile
network and WiBro which has control signal.
To support QoS assurance of VoIP in heterogeneous network,
network control platform is required by providing QoS
assurance by wireless resource control.
2.4 Requirements in service control layer
Service control platform is required for seamless voice
service in service control layer. When user terminal moves to
other network, service control platform interworks and
exchanges the service control signal and service profile
information between networks for seamless voice service.
2.5 Summary of requirements in each layer
Above mentioned requirements in three layers are
summarized as Figure 1.
F~~
2.3.1 Requirements of new IP based mobility function in
network layer
Owing to the insufficient mobility support of mobile IP,
simple IP, mobile SIP, and UMA, new IP based mobility
function is required for seamless mobility support in
convergence network. This new IP based mobility function
will be set-up as follows
* There is no need to change legacy network
* User terminal initiates handover technology by
reporting his mobility information to "Mobility Control
Platform".
* "Mobility Control Platform" updates and manages the
mobile and location information of user terminal
* When user terminal moves to new other network,
existing used channel is not disconnected until new
channel is connected using "Make before break"
technology
2.3.2 Requirements of mobility control platform in
network layer
To support QoS assurance of seamless mobility service in
heterogeneous network, mobility control platform is required.
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-.1.
i
Interworking of service control
signal between networks
Interworking of user service
profile information
i
Seamless handover support
QoS assurance for service
i
Multi-band multi-mode support
| | || || I T \ I User terminal identity and
authentication support
Figure 1 Summary of requirements in three layers
3. Service scenario for high quality mobile VoIP service
Service scenarios to provide high quality VoIP service are
composed of registration phase, service signal phase, real
service provisioning phase, and service closing phase.
3.1 Registration phase
Registration phase is composed of allocation phase of IP
address to user terminal and transmitting and saving phase of
SIP registration message concerning user profile. More detail
procedures of registration phase are as follows, and shown in
Figure 2.
1) Permanent IP address is allocated to originating user
terminal from mobility control platform
2) Local IP address (Network IP address) is assign to
originating user terminal from DHCP of local network
3) Received IP address from local network is reported to
Mobility Control Platform
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4), 5) SIP registration message is sent to HSS from
originating user terminal for user registration
6) Originating user profile information is saved in HSS
4) SIPntnVt interpeaon
X4W SxI r S
O
S
QOS et p nforion
Forw*ding
Nw Othei
Node I7
LoeWIAdd
IP
Local
F
3)Lo IP Address
Rep
Assign
ili%
8QOS SeI Entoremin
to NeiwrkfMode
4NeW Loa IPAddress
Repowt
Node 1
..0
c
Node 2
Figure 3 Overall procedure of service signal phase
Plaffnrm
3.3 Real service provisioning phase
1) Permanent IP Address
Allocate
Real service signal phase is composed of pre path set-up,
RTP session set-up, and bi-directional real data forwarding
phase. More detail procedures of real service provisioning
phase are as follows, and shown in Figure 4.
Figure 2 Overall procedure of registration phase
3.2 Service signal phase
Service signal phase is signal flow before real service
provisioning after registration phase, and composed of
allocation and reporting phase of new IP address,
interpretation and forwarding phase of SIP invite message,
sending phase of SIP 200 OK message. More detail
procedures of service signal phase are as follows, and shown
in Figure 3.
1) When user terminal moves to other network (domain,
subnet), new local IP address (New network IP
address) is assigned to user terminal from DHCP of
other network (domain, subnet)
2) New IP address received from other network is
reported to Mobility Control Platform, and Mobility
Control Platform manages location information based
on received IP address
3) SIP of originating user terminal is transmitted to
CSCF
4) CSCF interprets invited SIP from originating user
terminal
5) Invite message from originating user terminal is
forwarded to terminating user terminal
6) Terminating user terminal forwards "200 OK
message" to CSCF
7) CSCF forwards necessary QoS information for call
set up to Network QoS control platform
8) Network QoS control platform requests necessary
QoS information to network node
9) Originating user terminal receives "200 OK message"
from CSCF
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Mobj fyControl
PIa#eorrn
Network QS
Contro Ptor
r
4) SIP Registrafion.n
21 Local IP Addre
5) DestinamSlP Wilt Fwardeig
CSCF
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1) Network QoS control platform set-up the pre path
for QoS assurance of requested call
2) RTP session is set-up between originating node and
terminating node
3) Bidirectional real data is forwarded between
originating node and terminating node
Node 1
Node 2
Figure 4 Overall procedure of real service provisioning phase
3.4 Service closing phase
Service closing phase is composed of interpretation and
forwarding phase of SIP bye message and forwarding phase
of QoS releasing message. More detail procedures of service
closing phase are as follows, and shown in Figure 5.
1) Originating user terminal transmits "SIP bye
message" to CSCF
2) CSCF interprets "SIP Bye Message" received from
originating user terminal
3) SIP Bye message is forwarded to terminating user
terminal from CSCF
4) Terminating user terminal forwards "200 OK
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message" to CSCF
5) CSCF forwards necessary QoS information for call
release to Network QoS control platform
6) Network QoS Control Platform transmits release
message of allocated resources to network nodes
7) CSCF transmits "200 OK message" to originating
user terminal, and RTP (Real Time Transport
Protocol) session is closed
Overall procedure of service closing phase is shown in Figure
5
2) SIP Bye interpretation
CSCF
3) Destination SIP Bye Forwarding
5) QoS release Information
Forwarding
1) SIP Bye
Network QoS
Control Platlorm
6) QoS release Enforcement
7) SIP 200 OK
4) SIP 200 OK
to Network Node
Node I
Node 2
Figure 5 Overall procedure of service closing phase
4. Conclusions
Requirements in user terminal layer, network layer, service
control layer are mentioned, and service scenarios are
described in registration phase, service signal phase, real
service provisioning phase, and service closing phase to
provide high quality mobile VoIP service by supporting
seamless mobility in heterogeneous network among WiFi, 3G,
4G, and mobile WiMAX. Currently VoIP service with WiBro
and WiFi is under consideration, so proposed requirements
and service scenarios will became useful guidelines for high
quality mobile VoIP service.
REFERENCES
[1] ITU-T Recommendation Y200 1, General overview of NGN
[2] ITU-T Recommendation Y2201, NGN release 1 requirements
[3] ITU-T Recommendation Y20 12, Functional Requirements and
Architecture of the NGN
[4] ITU-T Recommendation Y2021, IMS for Next Generation Network
[5] ITU-T Recommendation Yrtconv, IMS based Real-time Conversational
Multimedia Service over NGN
ISBN 978-89-5519-136-3
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Feb. 17-20, 2008 ICACT 2008
Authorized licensed use limited to: Jyvaskylan Ammattikorkeakoulu. Downloaded on October 16, 2008 at 06:00 from IEEE Xplore. Restrictions apply.