age is defined that VoIP user’s packet error rate (PER)
has been above 2% due to packet loss and packet de-
lay exceeding the target budget. However, when this
channel-adaptive DB function is applied, it makes the
scheduler more flexible in choosing a BE traffic in
mixed traffic scenarios.
3 VOIP SERVICE OVER HSDPA
SYSTEM
3.1 Characteristic of VoIP Service
1) Traffic model and protocol: In this work, two traf-
fic models are considered for the cases of BE service
and VoIP service. In the context of BE traffic, we ap-
plied the full queue traffic model assuming that data
can be always sent when a queue of certain user is
chosen. On the other hand, conversation traffic can be
approximated to the two state Markov traffic model
with a suitable voice activity factor (VAF) (3GPP TR
25.896). The adaptive multirate (AMR) voice codec
is mandatory for voice service in HSDPA systems.
During bursts of conversation, with the AMR mode
of 12.2kbps, the VoIP application generates 32-bytes
voice payload at 20ms intervals (3GPP TS26.236).
During silent periods, a 7-bytes payload carries a si-
lence descriptor (SID) frame at 160ms intervals. A
typical VoIP protocol stack, which employs the real-
time transport protocol (RTP), is encapsulated to the
user datagram protocol (UDP). This, in turn, is car-
ried by IP. The combined these protocols demand a
40-bytes IPv4 header or a 60-bytes IPv6 header. Ob-
viously, the overhead caused from the header to sup-
port VoIP service seriously degrades the spectral ef-
ficiency. Therefore, efficient and robust header com-
pression (ROHC) technique can be used to reduce the
effect of relatively large headers in the IP/UDP/RTP
layers. This technique can reduce the size of the
IP/UDP/RTP headers as little as 2 or 4 bytes. Max-
imum compression 1 byte can be achieved by impos-
ing limitations (IETF RFC 3059, 2001).
2) Definition of VoIP capacity: The VoIP ca-
pacity is in the sense that there exists the maximum
number of VoIP users that can be supported per sec-
tor without exceeding a given outage threshold. In
packet-switch (PS) network, packets will be dropped
under network traffic loads congestion due to packet
loss and packet delay exceeding the target budget. Al-
though some packet loss occurs, the voice quality is
not affected if the amount of packet loss is less than
outage threshold. To proceed with this work, we as-
sume that the PER is kept within 2% and at least 95%
Table 1: Summary of end-to-end delay component for VoIP.
Delay component Delay assumption
Voice encoder 20ms(12.2Kbps)
NodeB scheduling+HARQ Max. 110ms
NodeB site 30ms
ROHC, RLC+MAC process
Downlink propagation
UE scheduling+HARQ 40ms
UE site 30ms
processing, buffering, etc
Uplink propagation
Backhaul delay 30ms
IP network delay About 42ms
of VoIP users should meet the above criterion (3GPP2
TSG-C.R1002-0).
3) End-to-end delay latency for QoS support: To
ensure end-to-end QoS, the low delay is one of the
most important criteria for maintaining high-quality
VoIP service. But, to attain high VoIP capacity,
the scheduler must have sufficient time to manage
voice packets. Of the assumed 285ms end-to-end de-
lay budget for qualified voice service, about 110ms
is available for scheduling in the downlink (ITU-T
G.114). The delay in IP and backhaul network is in
general bounded to 72ms (3GPP TR 25.853), which
is fixed value allowing us to focus on the delay bud-
get within radio access network as shown in Fig. 4.
The end-to-end delay budget in the case of mobile-to-
mobile conversation can be assumed as table 1. Al-
though VoIP performance depends on both downlink
and uplink performance, we would like to set aside
the consideration of both directions as comprehensive
study for future research.
3.2 System-level Simulation Setup
To investigate the performance evaluation with a
mixed voice and BE traffic, a system-level Monte-
Carlo computer simulation is accomplished in this pa-
per. The simulations are carried out with a regular
hexagonal 19 cellular model, where the distance be-
tween Node B is 1km. Mobile terminals should be
uniformly distributed on the 19-cell layout for each
simulation run and assigned different radio link mod-
els according to the assignment probability specified
in (3GPP TS25.101). Note that a realistic model of
the wave propagation plays an important role for the
significance of the simulation results. Shadowing is
modeled by a log-normal fading of the total received
power and a basic attenuation is determined by the
Hata model (3GPP2 TSG-C.R1002-0). Moreover, we
reserve the resources for control and common chan-
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