2015 Fifth International Conference on Advanced Computing & Communication Technologies

Solution to Reduce Voice Interruption Time during Handover of VoLTE Call in

Enhanced Single Radio Voice Call Continuity

Sumit Gautam

Dr Durgra Prasad Sharma, Professor

Sr. Technical Leader, LG Electronics

LG Soft India, Embassy Tech Square, ORR
Bangalore, India
sumit.gautam@lge.com

AMIT, AMU MOEFDRE and Adviser (IT) ILO &

UNDP,
dp.shiv08@gmail.com

Abstract LTE network is a packet switched based network,

which does not support circuit switched network feature like
voice call. Operators have been trying to deploy LTE network
in phases and in parallel they are supporting legacy networks
too. It is necessary to support voice call. There are many
approaches for supporting voice call to serve customer. Circuit
Switched Fall Back (CSFB), Simultaneous Voice & LTE (SVLTE), Over The Top solution (OTT), Voice over LTE via
Generic Access (VoLGA) and IMS based Voice over LTE
(VoLTE). SRVCC is a voice call handover process for an LTE
user, while it is in active IMS based voice session. In case of
SRVCC, control is performed by the home network which
causes voice interruption & call drops. To overcome these
issues ATCF and ATGW entities have been introduced. This
paper explains about the issues, causes and solution in SRVCC.

connection instead of a regular or legacy phone network.

IMS based VoIP call over LTE network (VoLTE) are the
way to support voice call over LTE (packet switched)
network. Voice call handover while user equipment is out of
LTE network area and other scenarios requires support of
SRVCC.
II.

Evolution in smartphone allows user for demanding more

data throughput and more user applications. Demand for
higher data rate, higher data consumption, and more users
interactive application forced operators, who were providing
2G or 3Gnetwork, to evolve & deploy LTE network. But
basic feature of mobile phone i.e. voice call must support in
any kind of network [8].

KeywordsLTE, VoLTE, SRVCC, eSRVCC, IMS,

I.

INTRODUCTION

The 3GPP Long Term Evolution (LTE) is a standardised

global network with high-speed, high-capacity data standard
for mobile devices. It is becoming commercialised in many
countries now. Many operators who are providing services
on circuit switched network are deploying LTE networks
too. The first fully commercial LTE network was deployed
in 2009. Combination of circuit switched telephony services
via CDMA and packet switched LTE data services are
present since 2010, where user equipment need to support
dual active radio to serve both services [ ]. By this
mechanism voice services via CS network and data services
via PS network have been served to user.

LTE network is an all Internet Protocol based mobile

network. It does not support traditional circuit switched
domain voice and requires supporting voice call. To support
voice call operators use various approaches including CSFB,
SV-LTE, OTT, VoLGA and VoLTE etc. In Circuit Switch
Fall Back (CSFB) voice traffic manage via legacy CircuitSwitched (CS) networks. Voice over IP (VoIP) over LTE
network (VoLTE) is a voice telephony solution based upon
IP Multimedia Subsystem (IMS) to deliver voice services
over LTE access. Single Radio Voice Call Continuity
(SRVCC) is a method for ensuring faster & reliable
handover of an LTE user, while it is in active IMS based
voice session. It is not an alternative of voice delivery but
rather a voice call handover process. SRVCC follows the
basic principle of IMS call control mechanism where
control is performed by the home network. Due to this
reason there are issues of delay in communication path
switching especially in case of roaming. Enhanced Single
Radio Voice Call Continuity (eSRVCC) supports control of
voice handover into visited network in such a way that the
visited network of the user equipment holds the voice data
and SIP signal paths at the time of the calling and receiving
of VoLTE calls.

CSFB was an evolution in earlier approach and it requires

single active radio to serve both services. LTE user
equipment using circuit switched fall back to support 2G/3G
legacy network systems are available since 2011. These are
less expensive, smaller, more power efficient single radio
solutions.
LTE network initially & mainly designed for data
network communication but its quality of service and
capacity allows operator & user to gain additional feature
e.g. high definition voice, enhanced video capabilities and
rich media communication etc. Voice over IP over LTE is
an IMS based telephony service which serves voice services
over LTE access network. It supports multimedia telephony

Voice over Internet Protocol (VoIP) is a technology that

allows to make voice calls using a broadband Internet
2327-0659/15 $31.00 2015 IEEE
DOI 10.1109/ACCT.2015.97

VOICE OVER LTE

627

based service e.g. HD voice, Rich Communication Suite

(RCS), instant video share etc. with the help of MMTel [5].
These are single radio solution with Single Radio Voice
Call Continuity (SRVCC). SRVCC seamlessly maintains
voice calls as mobile users move from LTE to non-LTE
coverage areas. Evolution on LTE voice is going on and this
evolution coming out in form of video call over IP, rich
communication services, interoperability across network
access methods and voice of WLAN etc.

to new network or it will be suspended until the UE returns

back to the LTE network. If the IP data transmission is also
handed over than it might operate in the lower speed of the
legacy network. The legacy network can also reject the
handover of IP data transmission if it is not able to process
it.

Quality of service (QoS) is utmost for satisfying user

experience especially while user is in voice call [4][6].
VoLTE quality of service measure include
Lesser voice interruption time during handover (0.3 sec)
Minimal call drop (less than 1%)
Lower packet loss (1% to 3%)

A. Circuit Switch FallBack (CSFB)

Figure 1. CSFB

CSFB is a procedure for switching to a radio access

system having a circuit switched domain, when a terminal
sends or receives a circuit switched communication such as
voice while camped on an LTE network [3].

CSFB incoming call scenario:

When incoming call arrives for LTE user, the call
request first arrives to MSC which to UE previously
registered to. When MSC gets the call request it sends
paging message to the related MME via SGs interface. This
message is forwarded to the UE which still connected to the
LTE network. If the user accepts the call it sends service
request to MME. The MME than notifies eNodeB to
transfer the UE to the legacy network. Then the eNodeB
decides the best network for the UE to perform the
handover.

Circuit switched fall back mechanism that uses the

legacy 2G/3G core network and radio access for providing
voice service to LTE subscribers equipped with dual mode
handsets. In Circuit switched fall back approach the LTE
user is handed over to all legacy circuit switched network
assuming it provides an overlapping coverage whenever a
request for voice call arrives. At the end of the call at the
legacy network the device may (but not necessary) reassociated & registered again with LTE network.

III.

To support Circuit Switched Fall Back (CSFB) a new

interface 'SGs' has been introduced connecting MSC (legacy
network) and MME (LTE network) as shown in Fig: 1. LTE
handset which CSFB is turned on, registered to both
networks the LTE and legacy network. The legacy network
needs to know the location information of LTE user to
enable first handover to legacy network when require.
Therefore the MME which tracks the location of the user at
LTE network continuously provides location information to
the legacy network MSC using the new SGs interface.
Assuming now that UE is initially served by the LTE
network as an active IP connectivity; when the UE decides
to make a voice call, it sends a service request message to
MME then MME checks if UE is capable of handling CSFB
and notifies eNodeB to transfer the UE to legacy network.
Before handing over the UE the eNodeB may request UE to
perform RF measurement of neighboring 2G/3G networks.
The eNodeB than decides the best network for the UE and
performs the handover. Any IP data transmission taking
place at that time voice call is placed will either handed over

SINGLE RADIO VOICE CALL CONTINUTIY

Single Radio Voice Call Continuity (SRVCC) system

has been introduced to perform seamless handover between
packet switched network and Circuit switched domain [1].
In SRVCC enabled systems UE does not require having two
active radios, but requires only a single active radio.
SRVCC is not an alternative of voice delivery but rather a
voice call handover process. SRVCC is a method for
ensuring faster & reliable handover of an LTE user, while it
is in active IMS based voice session. The challenge with
SRVCC is to make a handover while UE is connected to
only a single radio at given time as shown in Fig: 2 [9].
SRVCC is a handover mechanism from IMS based LTE
network to other networks in order to provide service where
LTE is not deployed. As an SRVCC-capable mobile
engaged in a voice call determines that it is moving away
from LTE coverage, it notifies the LTE network. The LTE
network determines that the voice call needs to be moved to
the legacy circuit domain. It notifies the MSC server of the
need to switch the voice call from the packet to the circuit

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domain and initiates a handover of the LTE voice bearer to

the circuit network as shown in the Fig: 12. Variations of
SRVCC support both GSM/UMTS and CDMA 1x circuit
domains.

introduced for Single Radio Video Call Continuity to

differentiate it from Single Radio Voice Call Continuity
(SRVCC).

Figure 4. SRVCC High Level Concept

Figure 2. SRVCC - Concept

When LTE signal strengths starts diminishing in the UEs

area, the UE signals to eNodeB about changes in signal
strength and the SRVCC handover is initiated. The eNodeB
than calculates the best network available to handle the
service of the served UE and sends the request of handover
to MME as shown in the Fig:4. In handover request eNodeB
also specifies the handover is SRVCC based. A new voice
call request is than send to IMS using special number
known as Session transfer number (STNSR) for SRVCC.
STNSR is unique number that is generated for each UE and
stored in the HSS. This number is send to the MME by the
HSS when UE first contact to the network. Receiving
STNSR number indicates SCCAS that the corresponding
call needs to be routed to a different network and it starts redirection process to a legacy endpoint. After the resources
preparation is complete MME confirms handover request
provided earlier by eNodeB. The eNodeB than transmits this
confirmation to the UE and also provide the required
information about the target network. In the final steps UE
is detected in legacy network and the call is re-established
with UE. Voice packet and non-voice packet can be
handover by this method. Although data rate is delimited by
is of the capability of legacy network.

To allow SRVCC continuity the UE and bus networks

LTE and the target legacy networks should support SRVCC.
In addition special interface Sv has been introduced
between the MME and MSC. For supporting SRVCC that
IMS network should also include an application server SCC
AS (Service Centralization and Continuity Application
Server). This application server and else signaling is
required for the process and shown in the Fig: 3.

Figure 3. SRVCC

Two versions of SRVCC are exists:

SRVCC handover from LTE to GSM or UMTS
networks defined by 3GPP
SRVCC handover from LTE to CDMA 1x network
defined by 3GPP2

Main issue observed in VoLTE and SRVCC is the voice

interruption time / jitter during handover cases. The
immediate solution to reduce the time came as
simultaneous redirection of the RAN and Session. By this
approach user experience has been improved and actual
interruption time is not unduly noticeable.

Voice call continuity between IMS over PS access and

CS access for calls that are anchored in IMS when the UE is
capable of transmitting/receiving on only one of those
access networks at a given time.

A. Voice Interruption Time

Voice interruption performance target is less than 0.3
seconds [4]. Inter RAT handover and session transfer are
responsible for voice interruption time as they break and
remake the connection, shown in the Fig: 5 and Fig: 6. This
can be improved by initiating these two procedures

Video call continuity from E-UTRAN to UTRAN-CS for

calls that are anchored in the IMS when the UE is capable of
transmitting/receiving on only one of those access networks
at a given time. In this specification, the term vSRVCC is

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simultaneously to run in parallel. Session transfer

procedure is quicker than inter RAT handover procedure.
Handover procedure delay includes duration between
receiving handover commands and synchronizing on the
new network & send confirmation. Total voice interruption
time is little larger than handover procedure time and
defined by the time from when the last voice packet is sent
over LTE until voice media is sent over CS access [7].

retention rate is the percentage of successfully completed

calls for a given user. Call retention target for legacy voice
is higher than 98% and SRVCC handover is more than 99%
at the time.
C. Service Continuity
Session transfer procedure and Access transfer procedure
are required for session continuity [2]. When an UE is active
in an IMS session, the Session Transfer procedures provide
service continuity between Access Networks and between
UEs having IMS subscriptions under the same operator. The
initial and all subsequent Session Transfer procedures are
initiated by the UE and are executed and controlled by the
same SCC AS. The SCC AS generates charging information
for all Session Transfers for an IMS session for the purpose
of billing and charging. The UE sends information required
by the SCC AS in order to execute Session Transfer
procedures.
IMS sessions from and to a UE are anchored at the SCC
AS in the home IMS and may also be anchored at the ATCF
in the serving (visited if roaming) network to provide
Service Continuity for the user during transition between
two Access Networks. Sessions are anchored at the SCC AS
in the home IMS, based on iFC. A 3pcc (3rd party call
control) function is employed at the SCC AS to facilitate
inter-Access Network mobility through the use of Access
Transfers between the two Access Networks. IMS media
sessions may be anchored by the ATCF during session
establishment depending on operator policy. Access
Transfers may be enabled in one or both directions as per
network configuration requirements. The SCC AS has the
capability to perform Access Transfers for a UE's sessions
multiple times. Initiation of Access Transfer procedures for
ongoing multimedia session may be based on the operator
policy received from the SCC AS [Fig: 7] [Fig; 8].

Figure 5. Voice jitter

IV.

ENHANCED SINGLE RADIO VOICE CALL

CONTINUITY

SRVCC follows the basic policy of IMS call control

whereby the control is performed in the home network. As a
consequences of this, especially in case such as roaming,
there used to be a issue because processing delay in
communication path switching are increased due to the fact
that call processing in the home network also becomes
necessary even for HO taking place in the visited network.
Therefore, an improvement was introduced into eSRVCC
whereby the control of voice HO can be confined in this
visited network in such a way that the visited network of the
mobile network anchors the voice data and SIP signal paths
at the time of the calling and receiving VoLTE calls. New
functional entities such as the Access Transfer Control
Function (ATCF) and the Access Transfer Gateway (ATGW)

Figure 6. Voice packet delay

B. Call Retention
SRVCC implementation objective was to reduce call
drop while users moving in and out of the LTE coverage. It
should support voice calls, emergency calls, and voice
handover to & from VoLTE and legacy networks. Call

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have been added in order to realize an anchoring inside of

the visited network [Fig: 7]. The function of the ATCF and
ATGW is to anchor the SIP signal and the voice media in
the visited network respectively. The ATCF is set to relay
the SIP signals beforehand within the IMS registration
procedure that precedes the launch of eSRVCC. In addition,
an ATCF identifier to identify the ATCF is sent to and
maintained in the HSS and MME beforehand. When
terminal launches a VoLTE call the ATCF that relays the
SIP signals Allocates an ATGW depending on the media
information, and immediately after the voice media is
anchored in the specified ATGW. Then when the eNode
finds that, based on the radio information etc., it is necessary
to switch the network used by UE1 from LTE to 3G, it
requests to reserve the necessary bearer resources in the CS
network [Fig: 8].
The Access Transfer Control Function (ATCF) is a
function in the serving (visited if roaming) network. When
(v)SRVCC enhanced with ATCF is used, the ATCF is
included in the session control plane for the duration of the
call before and after Access Transfer. The ATCF allocate a
STN-SR, include itself for the SIP session & instruct
ATGW to anchor media path based on operator policies.
ATCF keeps track of sessions (either in pre-alerting state,
alerting state, active or held state) to be able to perform
access transfer of the selected session. ATCF performs the
Access Transfer and update the ATGW with the new media
path for the (CS) access leg, without requiring updating the
remote leg. ATCF update SCC AS about access transfer and
also handle failure cases during access transfer [Fig: 10]
[Fig: 11].

Figure 8. IMS registration using ATCF enhancement

Figure 9. eSRVCC functional configuration & operational flow

V.

CONCLUSION

In this research attempts have been made to observe

issues in VoLTE especially in case of handover. It was
observed that Voice interruption time during handover; call
drop rate and session continuity were main obstacle to meet
up quality of service measures for VoLTE, while using
SRVCC handover mechanism. Introduction of ATCF and
ATGW entities in enhanced SRVCC made possible to have
simultaneous redirection of radio access network and
session. These two procedures run in parallel and reduce
voice interruption time & call drop rate even within
specified accepted limits i.e. voice interruption time during
handover must less than 0.3 sec and call drop rate must limit
within 1%. These QoS measures within limits satisfy user
experience and enable voice over LTE network with

Figure 7. IMS service architecture (ATCF enhancement)

The Access Transfer Gateway (ATGW) is controlled by the

ATCF and, if SRVCC enhanced with ATCF is used, stays in
the session media path for the duration of the call and after
Access Transfer, based on the local policy of the serving
network [Fig: 9].

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[2]

sufficiently good experience as well over circuit switched

network. Further, session transfer procedure has been found
to be quicker than inter radio access technology handover
procedure. Future work can be taken forward to reduce time
for inter RAT handover procedure, which can improve
overall voice interruption time.

[3]
[4]

[5]
ACKNOWLEDGMENT

[6]

We thank research supervisor and seniors of the

organization for their support & encouragement.

[7]

REFERENCES
[1]

[8]
[9]

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Figure 10. Session with PS media using ATCF (signaling & bearer path)

Figure 11. Session with CS media using ATCF (signaling & bearer path)

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Figure 12. Inter Rat Handover

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