Study on Appropriate‘ Voice Data Length of IP Packets
`for Voll’ Network Adjustment
`
`I-Iiroyuki OOUCHI, Tsuyoshi TAKENAGA' , Haj ime SUGAWARA, and Masao MASUGI
`NTT Network Service Systems Laboratories
`9-1 1, Midori-Cho 3-Chome, Musashino-Shi, Tokyo 180-8585, Japan
`‘Nippon Telegraph and Telephone West Corp.
`2-27, Nakanoshima 6-Chorne, Kira-Kn, Osaka-Shi, Osaka 540-851 1. Japan
`
`AbsmIct—Thls paper evaluates suitable voice-data length in
`1? packets for the adjustment of Voll‘ network systems. Based
`on measurements in a real environment, we examined the
`voice-quality level while varying the voice-data length of 11’
`packets under 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 voiee—quality
`level in a 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 (308 eonditl on: to
`achieve both high-transmission eificieney and stable voice
`quality.
`
`I. INTRODUCTION
`
`range of computer network technology has
`The
`expanded in size and diversity, and a wide variety of
`value-added applications for use on the Internet have
`appeared in recent years. Networks (eg. 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
`Voll’ (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 lP—based
`technology for computer-telephony integration, and this has
`resulted 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 sometimes difficuit.
`Because voice transmission on the Internet is unreliable,
`the current best-effort technology cannot guarantee the QoS
`or reliability of VoIP services. Various studies on Vol?
`technology have tried to establish reliable services and
`evaluate QoS properties [l-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
`code: 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 header‘ length 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 Protocol) connections for a voice
`gateway and decreased processing load for routing.
`The transmission cfficiency of VoIP can be enhanced by
`omitting redundancy in the
`IP/UDPIRTP (Real-time
`Transport Protocol) header information, compressing it by
`not resending header information that does not change afier
`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
`enhance the 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 [l0]) while changing
`the voice-data length of IP packets for various packet loss
`and jitter conditions. The above measure is widely used,
`and it 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 [1],
`l2]). 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.
`
`II. TRANSMISSION INEFHCIENCY OF VoIP
`
`the
`When packets transmit an analog voice signal,
`VoIP-GW (VoIP-Gateway) digitizes
`the voice signal
`through a codec, such as G.7ll (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 transmission efficiency.
`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-‘T803-7632-3I02:'$17.00 ©2002 [BEE
`
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`
`Apple 1015
`Apple 1015
`U.S. Pat. 8,243,723
`U.S. 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, so the ratio
`ofvoice data to the total is less than 25%
`
`_1L_h_y>t_e__sm
`
`aims
`
`14 bytes
`
`isisuouytes
`
`.........
`
`4
`
`"
`
`—'
`
`-"
`
`lbyies
`
`‘
`
`~.
`
`x
`
`‘u
`
`20 bytes
`
`8 bytes
`
`12 bytes
`
`Ghytes or more
`
`Fig. 1. VoIP packet structure for Ethernet.
`
`The voice-data length of an IP packet usually depends
`on the method used for the \bIP-GW. Eighty-byte voice
`data is often used for G.7l l, 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 set up.
`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 Vol? packet has overheads for the
`MAC header (in the case of the Ethemet), 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 Q05 in VoIP
`systems.
`
`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 reproduction rate of voice
`data in the worst case may be 36%. This explains why the
`VoIP quality decreases drastically with bit error rate. The
`above relationship is expressed by the following formula,
`
`Err_f=L*8*e
`
`(1)
`
`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.
`
`N8
`
`150
`
`kbtlsl
`
`OcoupIedbandwIdth(
`
`.3s2‘ rrirI
`
`5
`
`20 40
`
`60 80 100
`
`—O—G.7‘l1
`—I—G. 729
`
`Transmission cycle (ms)
`
`Fig. 2. Bandwidth occupied by Vo[P frames.
`
`C
`3 1
`‘E 0.5
`% O6
`5
`-
`o_4
`E 0.2
`E o
`
`—o—G 71
`1
`.
`-I-(3.729
`
`5
`
`so 100
`20 4o 60
`Transmission cycle (ms)
`
`Fig. 3. Transmission elflciency
`
`III.
`
`EXPERIMENT
`
`We evaluated the effect of changing the packet loss rate
`and the voice data length of IP packets on the setup
`described below. The background traffic generator fixed the
`frame length and the amount of background traffic.
`
`A. Test-bed network
`
`stxup we used to
`Fig. 4 shows the measurement
`evaluate the voice quality of a VoIP system.
`
`
`
`Fig. 4. Ted:-bed network.
`
`1619
`
`

`
`

`
`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
`flireshold, 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 less than a certain threshold, the
`VoIP-GW assigns long voice data. If 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
`\i'olP~GW
`
`ci.a.,gԤt,.,mai;. length Change voicedaia length
`
`Acquire network gndifim
`Change voicedala length
`
`Acquire network Edition
`
`Acquire;-etvaodt condition
`Change voioedaia length
`
`Acquire neimdt condition
`
`Fig. 8. Variable voice data length VoIP system.
`
`Next, we provide the threshold for network condition
`values (response time, packet loss rate and jitter 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.l14 [14] as follows.
`0 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:
`‘I60 bytes) ~ 160 ms (voice data length: 800bytes) under
`G.71l. Considering the delay time in the VoIP-GW, the
`threshold for the delay time in an IP network should be less
`than 200 ms.
`
`B. Packet loss rate
`
`From the results in Section III.C, the threshold for packet
`
` 20
`
`30
`
`ID
`
`Jitter average time (ms)
`(2) MeasuredPSQM+
`
`deviation
`Standard
`
`Jitter average time (ms)
`(la) Standard deviation
`Fig. 7. Examples ofmeasuredinfluence by jitter(G.'.l1l).
`
`From these results, we found there was a relationship
`between voice quality and voice-data length in VoIP
`systems as follows.
`
`TABLE I]
`
`Characteristics of voice-data length
`Lon voice data
`
`
`
`Tend to degrade
`More fluctuation in
`voice quality
`
`More fluctuation in
`voice uali
`High transmission
`efficien
`
`Occupied
`bandwidh
`Influence of
`
`packet loss
`
`
`
`Influence ofjitter
`
`
`
`
`
`Tolerant to packet
`
`loss
`Less fluctuation in
`Voice uali
`
`Less fluctuation in
`voice uali
`
`Stability of voice
`- uali
`
`
`
`
`
`
`
`
`IV.
`
`VARIABLE VOICE DATA LENGTH Vol? 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 Vol? 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 of the system works as follows:
`
`0
`
`network conditions
`system monitors
`The VoIP
`rate,
`jitter
`time) by
`(response time, packet
`loss
`periodically pinging the destination Vol?-GW after a
`call connection is set up. Here, we regard a late
`response time, high packet loss rate and a large jitter
`time as an inferior network.
`The VoIP-GW assigns long voice data to IP packets as
`
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