Study on Appropriate Voice Data Length of IP Packets
`for VoIP Network Adjustment
`
`Hiroyuki OOUCHL,Tsuyoshi TAKENAGA’, Hajime SUGAWARA,and Masao MASUGI
`NTT Network Service Systems Laboratories
`9-11, Midor+Cho 3-Chome, Musashino-Shi, Tokyo 180-8585, Japan
`'Nippon Telegraph and Telephone West Corp.
`2-27, Nakanoshima 6-Chome, Kita-Ku, Osaka-Shi, Osaka 540-8511, Japan
`
`Abstrect-This paper evaluates suitable voice-data length in
`IP packets for the adjustment of VoIP network systems, Based
`on measurements in a real environment, we examined the
`voice-quality level while varying the voice-data length of IP
`packets ander various network conditions. We found that a
`VoIP system with long-voice data has high-transmission
`efficiency but there Is a high deterioration in the voice-quality
`level in an inferior network, We also discovered that a VoIP
`system with short-voice data is tolerant to packet losses and
`preserves voice quality. Based on these results, we propose a
`VoIP system that sets the voice-data length of IP packets
`according to dynamically changing network QoS conditions to
`achieve both high-transmission efficiency and stable voice
`quality.
`
`1. INTRODUCTION
`
`The Tange of computer network technology has
`expanded in size and diversity, and a wide variety of
`value-added applications for use on the Internet have
`appeared in recent years. Networks (e.g. the Internet) have
`become
`broadband,
`and multimedia
`communication
`environments where data, voice and images
`can be
`exchanged have been rapidly improving. IETF is studying
`communication services that connect existing telephone
`networks with IP networks, and ITU-T is deciding on a
`VoIP (Voice over Internet Protocol) protocol and studying
`the technologies for communication services where INs
`(Intelligent Networks) can collaborate with IP networks.
`There has been remarkable progress in the IP-based
`technology for computer-telephony integration, and this has
`Tesulted in lowering communication costs. In particular,
`packet voice applications such as Netmeeting, CuSeeMe
`and other conferencing multimedia applications have
`become widespread.
`Due to the shared nature of current network structures,
`however, guaranteeing the quality of service (QoS) of
`Internet applications from end-to-end is sometimesdifficult.
`Because voice transmission on the Intermet is unreliable,
`the current best-effort technology cannot guarantee the QoS
`or reliability of VoIP services. Various studies on VoIP
`technology have tried to establish reliable services and
`evaluate QoS properties [1-4]. However, the QoS level of
`VoIP systems depends on many parameters including the
`end-to-end delay, jitter, packet loss in the network, type of
`codec used, length of voice data, and the size of the
`jitter-absorbing buffer (5), so that further investigations are
`needed to clarify the factors affecting QoS. For instance,it
`has been pointed out that the transmission efficiency of
`voice data carried by IP packets is not sufficient because of
`the high ratio of fieaderlength to voice-data length in VoIP
`packets. Hence,
`to provide efficient and reliable VoIP
`
`services,it is important to clarify the effect that changing
`the system’s voice-data transmission rate has, among others.
`Also, there are some other issues such as less degraded
`voice quality due to VoIP packet losses, decreased number
`of UDP (User Datagram Protoco]) connections for a voice
`gateway and decreased processing load for routing.
`The transmission efficiency of VoIP can be enhanced by
`omitting redundancy in the
`IP/UDP/RTP (Realtime
`Transport Protocol) header information, compressing it by
`not resending header information that does not change after
`call connection setup, and multiplexing the voice data of
`two or more call channels
`in one IP packet with a
`sub-header identifying call channels. In ITU-T and IETF,
`IP-based multiplexing methods have been studied to
`enhancethe transmission efficiency of VoIP [6-9].
`We evaluated a VoIP system with the intention of
`designing optimal network services for various network
`conditions. Based on measurements in a real network
`environment, we examined the voice-quality level (PSQM:
`Perceptual Speech Quality Measure [10]) while changing
`the voice-data length of IP packets for various packet loss
`and jitter conditions. The above measure is widely used,
`andit provides an objective quality level for the voice over
`existing telephone voice bandwidths (300 Hz-3.4 kHz), and
`it is recognized as an appropriate method with relatively
`little error to determine the subjective quality level (MOS:
`Mean Opinion Score {11, 12]). The smaller the PSQM
`value,
`the better
`the voice quality. Using the results
`obtained from our experiment, we created a new VoIP
`mechanism that enhances
`transmission efficiency and
`provides
`stable voice quality.
`It
`sets
`the appropriate
`voice-data length for IP packets based on the dynamically
`changing network QoS conditions.
`
`Tl. TRANSMISSION INEFFICIENCY OF VoIP
`
`the
`When packets transmit an analog voice signal,
`VoIP-GW (VoIP-Gateway) digitizes
`the voice signal
`through a codec, such as G.711 (64 kbps) and G.729 (8
`kbps), and it transmits voice data of a fixed length, where
`UDP and RTP are used to transmit voice data in real time.
`The structure of an IP packet over an Ethernet is shown in
`Fig. 1. Here the voice data length can be changed according
`to the VoIP transmissionefficiency.
`The MAC (Media Access Control) header, IP header,
`UDP header, RTP header and FCS (Frame Check
`Sequence) are necessary for transmitting voice data over
`the Ethernet, and preamble and IPG (Inter Packet Gap)
`should be considered as occupying bandwidth on the
`transmission line. For
`instance,
`the
`total occupied
`bandwidth is 98 bytes including IPG, preamble, MAC
`
`0-7803-7632-3/02/$17.00 ©2002 IEEE
`
`1618
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`Facebook Ex. 1015
`Facebook Ex. 1015
`U.S. Pat. 8,243,723
`US. Pat. 8,243,723
`
`

`

`header, IP header, UDP header, RTP header and FCS when
`transmitting 20-byte voice data. The 78 bytes
`thus
`correspond to the overhead of IP transmission, sotheratio
`ofvoice data to the total is less than 25%.
`
`___tabytes—
`
`O bytes
`
`tbytes
`
`46 1500 bytes
`
`dbyles
`
`En_f=L* 8 *e
`
`qd)
`
`where Err_f (%) is the errored frame rate, L (bytes) is the
`voice data length per IP packet, and e (%) is the bit error
`rate. Note that this does not take into consideration the
`possibility of bit errors in the packet header.
`
`ee
`
`we
`
`2 CF
`& 200
`% 150 N
`
`3
`
`5
`
`20 40
`
`f
`60 80 106
`
`Transmission cycle (ms)
`
`Fig. 2, Bandwidth occupied by VoIF frames,
`
`20 bytes
`S bytes
`i2bytes
`6 bytes ar more
`“Hoeees..{ G728
`& 100
`ghteeeeeeed
`Fig. 1. VoIP packet structure for Ethernet.
`f°) Salles,
`The voice-data length of an IP packet usually depends
`on the method used for the WIP-GW. Eighty-byte voice
`data is often used for G.711, whereas 20-byte voice data is
`used for G.729 in conventional VoIP communication.
`However, it is also permissible to change the voice-data
`length per packet by changing the VoIP-GW setup.
`Table I shows the relationship between the packet
`transmission cycle and voice-data length, and Fig. 2 shows
`the relationship between the packet transmission cycle and
`the bandwidth occupied by the VoIP frames of an Ethernet.
`The longer the transmission cycle becomes, the longer the
`voice-data length. Moreover, the longer the voice data in an
`IP packet becomes, the more the transmission efficiency
`increases because the VoIP packet has overheads for the
`MACheader(in the case of the Ethernet), IP header, UDP
`header, and RTP header (Fig. 3). However, the longer one
`packet becomes, the more packet errors are likely to occur,
`so it is
`important
`to evaluate how the network traffic
`conditions affect
`the packet behavior and QoS in VoIP
`systems.
`
`ec
`> 1
`£08
`= 0.6
`go
`BO.4
`5 0.2
`Eo
`
`—~eG71l
`~#—G.729
`
`5
`
`80 100
`60
`20 40
`Transmission cycle (ms)
`
`TABLE I
`
`Relationship between packet transmission cycle
`and voice data length
`
`Transmission
`
`A high traffic load can cause packet loss and jitter, and a
`poor transmission line can cause bit errors. Larger jitter
`than the jitter absorbing buffer size may cause packet loss,
`and bit errors may also result in packet loss if the data link
`layer
`(such as HDLC (High-level Data Link Control
`procedure)}) has a function for dropping irregular frames.
`Given 40 bytes of voice data in an IP packet, e.g., when the
`bit error rate is 0.01%, the reproduction rate of voice data
`in the worst case may be 96.8%. Moreover, given 800 bytes
`of voice data in an IP packet, the reproductionrate of voice
`data in the worst case may be 36%. This explains why the
`VoIP quality decreases drastically with bit error rate. The
`aboverelationship is expressed by the following formula,
`
`Fig. 3. Transmission efficiency
`
`I.
`
`EXPERIMENT
`
`Weevaluated the effect of changing the packet loss rate
`and the voice data length of IP packets on the setup
`described below. The backgroundtraffic generator fixed the
`frame length and the amount of backgroundtraffic.
`
`A, Test-bed network
`
`Fig. 4 shows the measurement
`
`setup we used to
`
`evaluate the voice quality of a VoIP system.
`
`Fig. 4, Test-bed network.
`
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`
`

`

`
`
`the measured PSQM+ and standard
`Fig. 6 shows
`deviation for five voice-data lengths (160, 326, 480, 640,
`800 bytes} given a constant packet loss rate of 3%. Here,
`the G.711 codec was selected and the eight voice-sample
`files were transmitted. The 160-byte voice-data packets had
`the highest
`tolerance to packet
`loss and the 800-byte
`voice-data packets had the highest fluctuation in PSQM+,
`indicating that the longest packet was affected most.
`
`160
`
`320
`
`480
`
`640
`
`800
`
`25
`
`2
`
`15
`
`+ S
`
`wyoa
`
`A voice quality tester measured the PSQM+ between two
`VolIP-gateways, and a network emulator varied the packet
`loss rate within a range from 0 to 5% and average jitter
`time within a range from 0 to 50 ms. Backgroundtraffic
`was generated in the same direction as the voice data
`packet flow
`.
`
`B. Voice samples
`
`The speech database [13] contained eight voice sample
`files (four samples from two men speaking in Japanese,
`four samples from two women speaking in Japanese),
`which was based on ITU-T recommendation P.80. The
`PSQM+ value in the results is the average of these eight
`samples. We also calculated the standard deviation from
`these.
`
`C. Results
`
`Voice-data length (bytes}
`
`(a) MeasuredPSQM+
`
`c 0.9£
`
`8o
`
`oD
`p 06avu&
`f)0
`
`Fig. 5 plots the measured PSQM+ for G.711 and G.729
`codecs versus packetloss rate, where we chose a maximum
`and minimum that the VoIP-GW could assign for each
`codec as the voice-data length of the VoIP packets. We
`concluded the following from these results:
`e
`Short-voice data and long-voice data have the same
`voice quality when packetloss rate equals zero.
`The higher the packet loss rate, the lower the voice
`quality (= higher PSQM+), and the longer
`the
`voice-data packet length, the lower the tolerance to
`voice-data packetloss.
`0.3
`actual
`for
`Considering that
`the PSQM+ value
`160 64)=800320480
`communications should be less than about 2.5, the
`G.7E1 codec can tolerate packet loss rates of up to 5%,
`and the G.729 codec can tolerate losses up to 2%.
`
`e
`
`
`
`
`
`Voice-data length (bytes)
`
`(b) Standard deviation
`
`Fig. 6. Examples of measured results for several voice-data length s
`(G.711, packet loss rate: 3%).
`
`the measured PSQM+ and standard
`Fig. 7 shows
`deviation influenced by jitter. Here, we also selected the
`G.711 codec. We concluded the following from these
`results:
`e
`There was no difference between long-voice data and
`short-voice data, though the more the jitter the lower
`the PSQM+ value, when jitter occurred in the
`network.
`Longer voice-data packets had higher fluctuations in
`PSQM+,showing that the longer packet was affected
`more.
`
`for actual
`the PSQM+ vaiue
`Considering that
`communications should be less than about 2.5, the
`G.711 codec can toleratejitter up to about 20 ms.
`
`Additionally, when we compared the PSQM+ values of
`jitter-absorbing buffers A (large) and B (small), we found
`that the jitter tolerance of the former was higher than that
`ofthelatter.
`
`3
`
`+ 25
`a 9
`®
`15
`
`—— 800 bytes
`—— 160 bytes
`
`01423 4
`
`5
`
`6
`
`Packetloss rate (%)
`
`(a) 160 and 800 bytes (G.711)
`
`PSQM+
`
`0
`
`1
`
`$2
`
`3
`
`4
`
`5
`
`Packet loss rate (%}
`(b) 20 and200 bytes (G.729)
`
`Fig. 5. Examples of measured PSQM+ for two-voice-data lengths.
`
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`
`

`

`
`deviation
`Standard
`
`Jitter average time (ms)
`(a) MeasuredPSQM+
`
`Jitter average time (ms)
`(b) Standard deviation
`Fig. 7. Examples of measured influence by jitter (G.711).
`
`From these results, we found there was a relationship
`between voice quality and voice-data length in VoIP
`systemsas follows.
`
`TABLE 0
`
`
`
`
`Characteristics of voice-data length
`Long voice data
`Short-voice data
`Wide
`Occupied
`bandwidh
` Influence of
`packet loss
`
`
`
`
`Influence ofjitter
`
`
`
`voice quali
`
`
`Tend to degrade
`Tolerant to packet
`
`More fluctuation in
`loss
`
`Less fluctuation in
`voice quality
`
`
`voice quali
`Less fluctuation ia
`
`
`
`Stability of voice
`quali
`
`
`efficien
`
`Iv.
`
`VARIABLE VOICE DATA LENGTH VoIP SYSTEM
`
`In the previous section, we obtained the characteristics
`of a VoIP system operating under
`inferior network
`conditions. They can be used to create a VoIP system that
`offers optimal voice quality and transmission efficiency by
`varying the voice data length in IP packets based on
`network QoS conditions. Fig, 8 shows the architecture of
`the variable voice-data-length VoIP system we proposed.
`The VoIP-GW ofthe system worksas follows:
`
`e®
`
`network conditions
`system monitors
`The VoIP
`rate,
`jitter
`time) by
`(response time, packet
`loss
`periodically pinging the destination VoIP-GW after a
`call connection is set up. Here, we regard a late
`response time, high packet loss rate and a largejitter
`time as an inferior network,
`The VoIP-GW assigns long voice data to IP packets as
`
`1621
`
`little delay is experienced in telephone
`long as
`communications, because longer voice data increase
`end-to-end delay. If the response time exceeds a certain
`threshold, the VoIP-GW reduces the voice-data length.
`If the response time is less than the threshold,
`the
`VoIP-GW increases the voice-data length.
`If the IP network is stable (packet loss rate of nearly
`0%),
`the VoIP-GW assigns long voice data. If the
`packet
`loss rate exceeds a certain threshold,
`the
`VoIP-GW reduces the voice-data length to preserve
`communication quality. The VoIP-GW then increases
`voice-data length when network conditions return to a
`stable state.
`
`If the jitter time is Ess than a certain threshold, the
`VoIP-GW assigns long voice data. ff the jitter time
`exceeds a certain threshold, the VoIP-GW reduces the
`voice-data length to preserve communication quality.
`The VoIP-GW then increases voice-data length when
`jitter time returns to a low level.
`VoIP-GW
`VolP-GW
`
`Change voice-data length
`
`Acquire network Sondition
`Changevoice-data length
`
`Acquire network condition
`
`Acquirenetwork condition
`Changevoice-data length
`
`Acquire network condition
`
`Changevoice-data length
`
`Fig. 8. Variable voice data length VoIP system.
`
`Next, we provide the threshold for network condition
`values (response time, packet loss rate andjitter time) to
`change the voice-data length. When network condition
`values exceed the threshold,
`the VoIP-GW varies the
`voice-data length and jitter absorbing buffer size.
`
`A, Response time
`
`ITU-T defines the guidelines for one-way transmission
`time in G.114 [14] as follows.
`8 to 150 ms: Acceptable for most user applications.
`150
`to
`400 ms: Acceptable
`provided
`that
`Administrations are aware of the transmission time
`impact
`on
`the
`transmission
`quality of user
`applications.
`above 400 ms: Unacceptable for general network
`planning purposes; however, it is recognized that in
`some exceptional cases this limit will be exceeded.
`
`The delay time for the source and destination VoIP-GW
`used in our evaluation was about 90 ms (voice data length:
`160 bytes) ~ 160 ms (voice data length: 800bytes) under
`G.711. Considering the delay timé in the VoIP-GW,the
`threshold for the delay time in an FP network should be less
`than 200 ms.
`
`B. Packet loss rate
`
`From the results in Section III.C, the threshold for packet
`
`

`

`loss rate should be about 5% under G.71 1.
`C. Jitter time
`From the results in Section IILC, the threshold for jitter
`time should be about 20 ms under G.711.
`
`V. BEHAVIOR OF THE PROPOSED VoIP SYSTEM
`
`VI.
`
`CONCLUSION
`
`In this paper, we introduced some problems with VoIP
`systems and some methods ofefficiently transmitting VoIP
`packets by multiplexing transmission and other means.
`We evaluated the voice quality of a VoIP system while
`changing the voice-data length under various network
`conditions in a real network environment. We found that a
`VoIP system with short voice data in IP packets is superior
`to one with long voice data in terms of tolerance to packet
`losses and the stability of voice quality. To solve the
`problemsofinefficiency in a VoIP system, we proposed a
`new system that changes the voice-data length in IP packets
`based on dynamically changing network QoS conditions.
`This system achieves both high transmission efficiency and
`stable voice quality. We also described the behavior and the
`influence on IP networks of varying voice-data length
`under changing network conditions, and demonstrated the
`effectiveness of the proposed system.
`
`REFERENCES
`
`[2]
`
`[3]
`
`{4]
`
`[5]
`
`[6]
`
`[8]
`
`[9]
`
`[0]
`
`[k1]
`
`(f2]
`
`[£4]
`
`Before changing voice-data length
`o0000 eat 0000
`00000
`00008
`=e (BES
`
`( WOropped
`=
`
`O Dropped
`oO
`After changing voice-data length
`
`We studied how the relay node (e.g. router) of an IP
`network was affected by the VoIP-GW behavior above.
`Under stable network conditions, because the long voice
`data of a VoIP packet is assigned, transmission efficiency is
`high, and excess bandwidth can be used for non-voice data
`communications, The relay node incurs lower processing
`loads for long voice-data packets than for short voice-data
`packets.
`Whenthe relay node switches IP packets, it accumulates
`IP packets in its buffer, then routes them to a line. If there
`ate too many processed packets and the buffer is full,
`packets overflowing from the buffer may be dropped. As
`the processing load for the relay node becomes larger and
`the congestion level of the IP network increases, packet
`loss occurs. Then, the VoIP-GW assigns shorter voice data
`(1] M. Masuda, and K. Ori, “Network Performance Metrics in
`and WolP packets increase. However,
`the shorter data
`Estimating The Speech Quality of VoIP ”, Proc. ofAPSITT2001, pp.
`means that the lost voice data wili be distributed among
`333-337, Nov. 2001
`many voice data packets. Thus, there is less lost voice data
`L. Zheng, L. Zhang, D. Xu, “Characteristics of Network Delay and
`Delay Jitter and its Effect on Voice over IP (VoIP)’, Proc. of ICC
`in the short packets than in the long packets even if the
`2001, pp. 122-126, 2001
`packet loss rate becomes higher as a result of using the
`L. F. Sun, G. Wade, B. M.Lines, E. C. [feachor, “Impact of Packet
`shorter packets.
`Loss Location on Perceived Speech Quality”, Proc. of IP Telephony
`the amount of
`In current network environments,
`Workshop 2001, Apr. 2001.
`non-voice data transmitted over IP networks exceeds that
`[http:/Avunwfokus.gmd.de/events/iptd2001/]
`§TU-T Recommendation, G.114: “One Way Transmission Time”,
`of voice data. Therefore, shortening the voice data of the
`1996,
`VoIP packets has little effect on congestion, because the
`M.H. Sherif, A. Crossman, “Overview of Speech Packetization”,
`number of non-voice data packets basically determines the
`TEEE Symposium on Computers and Communications, pp. 296-304,
`1995
`level of congestion. Thus in most cases,
`a variable
` S. Casner and V. Jacobson, “ RFC2308: Compressing [P/UDP/RTP
`voice-data length VoIP system is useful for ensuring voice
`headers for low-speed serial links,” IETF, Internet Draft, Feb. 1999.
`quality (Fig. 9(a)}. If, however, the IP network carries only
`[7]|K. El-Khatib, G. Luo, G. Bochmann and F-. Pinjiang, “ Multiplexing
`Scheme for RTP Flows between Access Routers,” IETF, {nternet
`voice data, shortening the voice data length per packet will
`cause more packet loss than in the mixed packet case, so
`.
`Draft, draft -ietf-avt-multiplexing-rtp-01 txt, Oct. 1999.
`ITU-T Study Group 15, Delayed contribution D.510 (WP2/15), “A
`our VoIP system may be limited for use in networks
`Result ofPerformance Evaluationof IP -based Circuit Multiplexing
`carrying mainly voice data (Fig. 9%b)).
`Schemes,” Geneva, Jun. 1999.
`ITU-T Study Group 15, Temporary Document TD.24 (WP2/15),
`“Baseline document of Rec. G.ipome (IP-based Circuit Multiplexing
`Equipment),” Geneva, Feb. 2001
`ITU-T Recommendation P,861, “Objective Quality Measurement of
`Telephone-band (300-3400 hz) speech codecs”, Aug. 1996.
`ITU-T Recommendation, P.800: “Methods for Subjective
`Determination ofTransmission Quality”, 1996.
`ITU-T Recommendation, P.830: “Subjective performance
`assessment of telephone-band and wide-band digital codecs”, 1996.
`[13] NTT-AT CD-ROM, “Multilingual Speech Database for
`Telephonometry £994”, 1994.
`ITU-T Recommendation, G.114:" One way transmission time”’,
`1996
`
`oe an HOD og eeeal goono
`
`008 GE) vDropped
`
`O00 GOGO y Orpped
`
`O b-ong-voice-data packet
`0
`Shortvoice-data packet
`(a) Mixed packet case (useful)—(b) Only voice packet case (not useful)
`
`E] Non-voice-data packet
`
`Fig. 9. Comparison of dropped-packet behavior.
`
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