2012 2nd Baltic Congress on Future Internet Communications
Voice and video streaming in wireless computer networks - evaluation of network delays
Damian Bulira
Krzysztof Walkowiak
Department of Systems and Computer Networks Wroclaw University of Technology Wroclaw, Poland e-mail:
[email protected]
Department of Systems and Computer Networks Wroclaw University of Technology Wroclaw, Poland e-mail:
[email protected]
Abstract
-
End-user mobility and multimedia streaming
are current trends in Internet growth. Wireless technologies besides of their constraints - need to meet user requirements and even forecast them. Nowadays, it is very common to watch live TV program on a cell phone or on a screen inside the bus. Websites that allow sharing of audio or video transmission are at the top of the bandwidth usage ranks. Taking under consideration those issues associated with mobile access, which is almost as efficient as wired connections, we get enormous amount of information, which is available everywhere around the
world.
In
this
paper,
we
compare
wireless
access
technologies that are commonly available in Poland. The first group includes wireless LANs, that are still widely used as the "last
mile"
access
links
in
rural
environment
or
inside
buildings. The second group contains cellular packet access technologies developing from GPRS up to HSPA+ nowadays. The authors present theoretically possible data transfers and compare them against results obtained by testing existing networks. Moreover, we evaluate jitter and average delay in wireless
connections
evaluation
shows
that
during
streaming.
wireless
networks
Real
networks
generally
allow
streaming with no quality loss. However, a large gap is visible between first popular 3G standard - UMTS R99 and its enhancements - HSDPA and HSUPA. The newest available technologies and a future vision of LTE and 4G networks will allow every user to use mobile high definition streaming media anywhere.
Keywords-network delay, wireless technologies, conjerencing, video conjerencing, streaming I.
voice
INTRODUCTION
Currently access technologies based on wireless medium are becoming a more and more popular way of Internet access. One of the causes of this fact is a constant development of cellular technologies. Wideband connections are available for the vast majority of mobile phones users. Smartphones market, where Internet access become even more desirable is emerging rapidly. Huge amount of information that is intended for mobile end-users also
978-1-4673-1671-2/12/$31.00 ©2012
IEEE
156
promotes a development of wireless technologies. Streaming services like mobile TV, radio, music or voice and video conferencing need to be delivered smoothly to each user situated literally in every place on the planet. Cisco predicts that Internet video will reach 62 percent of consumer Internet traffic by the end of 2015, not including the amount of video exchanged through P2P file sharing. The sum of all forms of video (TV, video on demand (VoD), Internet, and P2P) will continue to be approximately 90 percent of global consumer traffic by 2015 [1]. In the nearest future with a growth of mobile technologies it could be assumed that this type of connection will become the most popular one. There are forecasts (e.g., in [1]) that traffic from wireless devices will exceed traffic from wired devices by 2015. Wireless access is not only used for peoples' convenience. It does not require any cable infrastructure so it may be easily used in places where maintaining conventional links is not possible or economically unjustified. Those may be temporary sites or rural environment. Here besides of cellular connections wireless LAN solutions may be used. WLAN are also the most popular technology applied inside buildings. Bringing together wireless access and streaming we encounter few crucial problems. The most important ones are network latency and bandwidth constraints. Because of wireless medium characteristic delay is the major problem in this type of connection. Streaming services are extremely sensitive to delay variations Gitter) and interactive services such as conferencing requires low delay to deliver a good quality of experience (QoE) [2] for the end-user. Taking under consideration all of those facts mentioned earlier, we make comparative performance analysis on streaming in various types of popular "last mile" wireless connections. Main contributions of this paper are 1) A comparison of wireless access connections on bandwidth and delay characteristics and 2) Performance tests of streaming services quality in those networks. This paper is structured as follows. In section 2 we present available wireless access technologies and their short characteristics. Then, in section 3 streaming services architecture is briefly introduced. Next, in section 4 we focus on delay components in streaming services. Section 5
describes the methodology of performance evaluation. In tlI section 6 we present test results and in the last 7 section the authors conclude their work. II. A.
WIRELESS NETWORKS
Wireless LANs
The most popular wireless LAN access technologies used as "last mile" links are IEEE 802.11 a, b and g networks. They operate in ISM (Industrial, Science, Medicine) band, that makes them widely accessible for users. Available bandwidth in all standards is sufficient for commonly used streaming services. An important problem is delay, that is caused mostly by CSMA/CA medium access mechanism [3]. When the medium is highly occupied or significant noise level is detected, CSMNCA prevents transmitter from sending a frame and runs the backoff algorithm, that may cause delay, especially when the medium usage is high. It is important that IEEE 802.11 b and g standards operate in 2.4 GHz band. This spectrum is widely used not only in wireless networking but also other technologies such as Bluetooth, cordless phones or wireless CCTV cameras [4]. Because of that, outdoor connections more often use 802.11a mode that uses 5 GHz band - also unlicensed, but with greater number of channels and smaller number of noise sources. Comparison of basic WLAN parameters is shown in Table I. B.
Cellular networks
Generally, currently available data transfer technologies nd 2 in cellular networks can be divided into two groups generation networks (2G) and 3G networks. Still there are a lot of places were only 2G network is available - it concerns mostly low populated areas and is economically justified. In this paper, we compare following cellular technologies: • 2G: o General Packet Radio Service (GPRS) o Enhanced GPRS (EDGE) • 3G o Universal Mobile Telecommunications System (UMTS) o High Speed Downlink Packet Access (HSDPA) o High Speed Packet Access (HSPA) o Evolved HSPA (HSPA+) Data rates in each cellular technology vary due to different coding schemas or channel/timeslot usage. _
TABLE!.
Ba nd Ch a nnel leng th Ava ila b le ch a nnels Modulation
BASIC WLAN PARAMETERS
S02. lla 5 GHz 20 MHz 19 OFDM
S02. llb 2,4 GHz 20 MHz 13 HRlDSSS
S02. llg 5 GHz 20 MHz 13 OFDM,HRlDSSS
157
GPRS is the first packet data transmission technology. Maximum one way data transfer is 80 kb/s (with use of 4 timeslots and CS-4 coding scheme). It is also the first technology that enables the service provider to charge user for data transfer and not for transmission length. EDGE is next end last step in evolving 2G packet services. Maximum transfer speed is 296 kb/s. Bandwidth increase is possible due to more sophisticated modulation and new multislot classes. 3G networks were designed to meet a need of new multimedia services transmission. Currently the most rd popular 3G standard is UMTS, developed by 3GPP (3 Generation Partnership Project) consortium. The first standard - UMTS Release 99, introduced in 1999 is based on WCDMA (Wideband Code Division Multiple Access) technology and allows the user to transmit data up do 384kb/s in both directions. With 3GPP R5 specification HSDPA was introduced. It enhances downlink speed up to 7.2Mb/s. HSPA is simply HSDPA and HSUPA (High Speed Uplink Packet Access) joint together. It enables uplink transmission up to 5.7 Mb/s. HSPA+ is currently the 3G standard that provides the highest data rates with 27.95 Mb/s downlink and 11.5 Mb/s uplink rate. In HSPA+ MIMO (Multiple Input Multiple Output) technology is introduced. Due to multiple antenna transmission, high data rates are achieved using modulations that are less complicated, as well as less vulnerable to interferences. In comparison to WLANs the latency is even a bigger problem in cellular networks. Total latency is a sum of a delay on a link between end-user and base station and a delay in operators network [5]. Previously it was investigated in [6] and this work provided some real-test results of cellular networks delay (Fig. 1). III.
STREAMING SERVICES
Most common streaming transmISSions that require significant link capacity are voice and video transmissions. Streaming media is delivered directly from a source to a destination in real time. It may be compared to TV signal transmission, but in contrast to TV broadcasts, traffic in IP networks can be transmitted in two ways. That provides an interaction between sender and receiver and introduces such services as video on demand, voice calls and video conferencing. Most of streaming is based on UDP transmission. This protocol does not support error correction and even small packet losses may result in notable quality issues. On the other hand there are researches that propose to use TCP in streaming, but without special algorithms that handle retransmission handling a delay may increase significantly [7]. Every streaming transmission system consists of following parts: • Capturing and coding device • Data server • Distribution and delivery network • Player
700
�-----
Fs
-
frame size in bits
Ls
-
link bit rate
600 500
].
400
�
300
200 100 a GPRS
EDGE
Fig ure 1.
UMTS R99
HSDPA
HSUPA
LTE
La tency in cellula r networks
It this paper we focus on the distribution and delivery network. The connection between a client and a server is maintained as long as IP communication between them exists. To provide good quality of streaming, some link parameters must be maintained, what we analyze in following parts of this work. In a network, the most interference vulnerable part is the "last mile" segment and the authors will continue analysis on behavior of network parameters during transmitting streaming services in wireless "last mile" links.
IV.
LATENCY IN STREAMING SERVICES DELIVERY
With constant growth of wireless networks capacity, it is essential to take under consideration the latency. Apart of latency in delivery networks, during multimedia streaming there are other components that belong to total end-to-end delay [8]. Particularly in streaming we distinguish following delays: a) propagation delay b) serialization delay c) queuing delay d) de-jitter buffer delay (playback delay) e) coding delay f) packetization delay Since we focus on delay in wireless networks, not on the delay introduced by streaming codecs, we will analyze a), b) and c). Propagation delay is a time that radio wave travels from a transmitter to a receiver. On short distance links which we are considering, this time may be omitted : Serialization delay is a time that transmitting device needs to introduce the whole packet into the medium. This delay can be calculated from (1).
For low speed and large frame size it has a visible impact in total delay calculation, but in most commonly used wireless technologies the propagation delay is less than I ms, which has insignificant impact on quality of transmission. The most crucial delay factor for transmission in wireless networks is queuing delay. It occurs when an interface is busy due to transmission of other frame. Then the frame is buffered and the time that it spends there is queuing delay. Statistically most of the packets will not be queued if link capacity is occupied in less than 50%. For wide bandwidth links queuing delay is not significant until the link is not saturated, what occurs when it is utilized in 96-97% (Fig. 2). Then delay is growing exponentially and packet loss occurs. As part of the queuing delay, we also consider a delay that is caused by the media access method. For example, in 802.11 networks CSMA/CA mechanism introduces a very large delay when the access point is highly occupied or noise level is high. When the transmitter detects that medium is busy, backoff algorithm takes place and transmission is on hold. Then it tries again, in case of another failure backoff time is longer, that causes significant delay when medium is highly used. ITU-T provides G.114 recommendation [9], that includes guidelines of upper end-to-end delay boundaries in voice transmission. Recommended delay value under which users experience good quality call is 150 ms. Anyway there are applications like videoconferencing systems or other with high interactivity level, that require delay under 100 ms. Document mentioned above sets 400 ms boundary as maximum delay above which connection quality is unacceptable.
70,0
35,0
60,0
30,0
-f
50,0 "'
!. ;;-
�
....I
40,0 30,0
- �
20,0 10,0 0,0 0
5000
10000
15000
20000
25000
.,.11
30000
25,0 20,0 7
E
15,0 10,0 5,0 0,0
35000
40000
Bitrate [kb/s] -
(1) Fig ure 2.
where: Ds
-
serialization delay
158
latency
- packet loss
Network dela y in b a ndwidth function during video tra nsmission
� i
n.
V.
TESTBED
Both benchmarks of WLAN and cellular technologies were performed on real networks. Providing reliable test results on wireless networks is not an easy task due to characteristics of the wireless medium. Interferences from other devices working in the WLAN spectrum might be significant a similar situation occurs in cellular connections. We used public networks, so actual capacity of base station and accurate distance to it is not given. Trying to provide as accurate results as possible, we performed our tests multiple times, we ran the same scenarios in different cellular networks and averaged the results. The testbed setup is shown in Fig. 3. The streaming server was situated in Wroclaw Center for Networking and Supercomputing [10] data center that is directly connected to PIONEER [11] network. This minimizes the network delay and jitter and avoids bandwidth problems. During testing WLAN networks, we plugged access point directly to the core network, unfortunately this setup is not possible in cellular networks, but connection from PIONEER network to cellular operators is reliable and fast. To simulate different streaming possibilities on the server side we used MyConnection Server [12] application that simulates network streaming. This tool is highly customizable and it is possible to simulate various types of streaming traffic. Client uses a web interface to communicate ad exchange data with the server. The benchmark consisted of RTT, maximum bandwidth, jitter and packet loss tests. During performance tests simulation of voice call with G.729 codecs and video transmission with 256 kb/s, 512 kb/s and 1024 kb/s bitrate was performed. Video streaming transmissions were tested one way - from the server to the client (half duplex - HDX), simulating streaming such as VoD. Both voice and video streaming were tested in two way transmISSIon (full duplex - FDX), simulating voice/video conference. Total test set includes 10 different streaming variants for 6 types of cellular networks and 3 WLANs. Additionally, each test performed on cellular networks was made using 3 different operators (Polkomtel, P4 and PTK Centertel). Tests that required more bandwidth than a particular technology provides were omitted. VI.
s------
Client (player)
Base station
Fig ure 3 .
Streaming server
Testb ed setup
The measurement of latency from the server to the client is a hard task due to the clock synchronization problem. In this paper the authors measured RTT and assumed half of this metric as approximated one-way delay. Test results are shown in table II. Measured delay values are close to minimal theoretical values for each link. In Fig. 4 on a logarythmic scale we can see that GPRS latency deviates from rest of technologies. It may be concluded that interactive transmission for video is impossible and voice call would have poor QoE. Of course wireless networks have visibly smaller delay than cellular ones. Moreover, the latency in wireless networks is very close to latency in wired connections. Results of bandwidth tests are shown in Fig. 5. We can observe development of cellular technologies. Available bandwidth in HSDPA is far distanced from maximum offered by this technology. During tests it did not achieve half of the maximum speed (3,6 Mb/s) what refers to category 5 and 6 in HSDPA connection. Visible is also the asymmetric characteristic of most cellular connections (apart of symmetric UMTS R99) and equal rates in both directions for WLAN. TABLE II.
TEST RESULTS OF BANDWIDTH AND RTTMEASURMENTS
b a ndwidth [kb /s1 uplink Downlink 21 45 104 220 371 290 2412 240 6270 1556 2089 83 69 21826 20943 5122 6029 1948 1 18700
GPRS EDGE UMTS R99 HSDPA HSPA HSPA+ 802. lla 802. llb 802. llg
RTT 59 0 180 117 121 126 113 2,6 2,8 2,5
TEST RESULTS 1� ,--
The first test was to measure the link bandwidth and round-trip time (RTT). Data was transmitted separately from the server to the client and in the reverse way. We measured only data rate available in the application layer (without overhead). That helps to assess the available bandwidth for particular streaming service. RTT provides information about total time that packet needs to travel from the client to the server and backwards. This provides estimated one way link latency. Due to the asymmetric link characteristics, half of the RTT time cannot be assumed as one way delay.
100
10
GPRS
EDGE
Fig ure 4.
159
UMTS R99
HSDPA
HSPA
HSPA+
802.11a 802.11b 802.11g
RTT results in wireless connections
TABLE IV.
25000 ,------
� -"
�]
20000
+-----
15000
+-------
HDX
10000
5000
+-------_.1--111--
GPRS
EDGE
UMTS R99 HSDPA •
Fig ure 5.
HSPA
downlink
_
HSPA+
802.11a 802.11b 802.11g
uplink
Ba ndwidth test results
The next step is jitter and packet loss measurement during a voice call. First, we simulated G.729 call that uses 8 kb/s bitrate. Simulation results are shown in Table III. The transmission should be possible in each of presented wireless technology. Anyway, the results show that in the case of GPRS and UMTS the transmission jitter is too high to provide good call quality. That large delay in GPRS was predictable, but in 3G technology it is unacceptable, minimizing this delay using large playback buffer would introduce a delay that in connection with other delay sources will result in very poor voice call quality. Other wireless technologies pass the voice call test without any problems. Next tests concern video transmission both as one-way streaming from server to client and full duplex transmission (simulation of video call). We simulated 3 bitrates of video transmission (256 kb/s, 512 kb/s and 1024 kb/s). The 256 kb/s video streaming exceeds the possibilities of GPRS and EDGE connections (table II), results of the tests in the other technologies are shown in Table IV. HSDPA as it was mentioned before enhances only downstream link, uplink still relies on UMTS Release '99, so two-way connection presents poor quality. Apart of that this transmission was possible in every other network.
TABLE III.
JITTER AND PACKET LOSS DURING VOICE TRANSMISSION (CODECG.729)
Jitter [ms] downlink uplink G PR S
14,3
24S,6
JITTER AND PACKET LOSS DURING VIEO TRANSMISSION (256 KB/s BITRATE)
0,2
EDGE
3,7
5,6
0
0
UMTS R99
2,2
92,9
0
0
HSDPA HSPA HSPA+
4,1
3,6
0
0
3,9
9,S
0
0
3,9
3,5
0
0
S02.lla
3,9
