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video over IP
Computer Communications 25 (2002) 863±873
www.elsevier.com/locate/comcom
Effect of delay and delay jitter on voice/video over IP Liren Zhang*, Li Zheng, Koh Soo Ngee Network Technology Research Center, School of Electrical and Electrical Engineering, Nanyang Technological University, Singapore, Singapore 639798 Received 28 August 2001; accepted 3 September 2001
Abstract Internet, especially in the situation when network has bursty background traf®c. However, it is usually dif®cult to evaluate the statistics of delay and delay jitter in such bursty traf®c environment due to their transient queuing nature. This paper focuses on the performance behavior of delay and delay jitter at IP packet level and also their effects on the performance of voice and video packet streams when they are multiplexed with bursty background traf®c in IP routers. The performance analysis of delay and delay jitter is done using the transient queuing solution technique. The numerical results are presented in terms of probability density functions (pdfs) of delay and delay jitter, which focus on their effects on speech samples and video frames. q 2002 Elsevier Science B.V. All rights reserved. Keywords: Voice over IP; Video over IP; QoS; Delay; Delay jitter
1. Introduction In today's networking, transmission of voice and video over the Internet are attractive alternatives both to conventional public telephony and leased lines. Voice over IP (VoIP) refers to real-time delivery of voice packets across networks using the Internet protocol. An appeal of VoIP is based on the capability to facilitate voice and data convergence at application layer. However, this integration effectively ®lls up the data communication channels and ef®ciently provides bandwidth consolidation. It is more important that the Public Switched Telephone Networks (PSTN) toll services can be bypassed using the Internet backbone to slash in prices for long distance calls [2]. On the other hand, video over the Internet has also being seen as the ideal last-mile solution for cable modem, DSL, and wireless networks since it allows service providers to bundle their offerings. Voice and video services often have more restrictive QoS requirement on delay and delay jitter than data transfer applications have, one of the typical problems with the transmission of voice packets and video packets over IP is the QoS guarantee. In fact, the QoS for such video or voice services can be affected by delay, delay jitter and unreliable packet delivery, which are the typical characteristics of the basic IP-network service, because packets are naturally transported using the best effort mechanism on ®rst-come, * Corresponding author. Fax: 161-792-0415. E-mail address: [email protected] (L. Zhang).
®rst-served (FIFO) basis. On the other hand, the size of data packet is variable in nature that allows large data ®le transfers to take the advantage of ef®ciency associated with larger packet sizes. However, these characteristics may contribute large delays and large delay jitters to packet streams, which are the most important concerns of the QoS for voice and video over the Internet. If the network is not properly controlled to meet the QoS requirements, the quality of voice and video is certainly affected. This is particularly true when voice or video packets are transmitted over the public Internet, where voice or video users have few options to secure the end-to-end QoS in terms of delay and delay jitters. ITU-T Recommendation G.114 [9] speci®es that one-way transmission time for connections with adequately controlled echo of voice should be ranging from 0 to 150 ms to be acceptable for most user applications [7]. However, video streaming over IP is able to handle a large jitter margin up to 2 s by using playback buffer at the receiver end. For an end-to-end VoIP call connection, three types of delay are experienced [3]: accumulation delay, processing delay and network delay. The accumulation delay is ranging from 125 ms to several milliseconds that are caused by the processing of voice samples. The processing delay is caused by the packetization of voice samples. The network delay is caused by multiplexing and buffering of voice packets when they are transmitted across networks. Delay jitter is another important parameter that should be taken into account in the support of voice and video services over IP networks. When voice or video packets are
0140-3664/02/$ - see front matter q 2002 Elsevier Science B.V. All rights reserved. PII: S 0140-366 4(01)00418-2
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Fig. 1. Queuing model for delay jitter.
transmitted over the Internet, packets may experience variable delay, called delay jitter. For non-real-time data transfer applications, such as ftp and telnet, delay jitter has little real impact on QoS. By contrast, real-time applications such as audio, video and voice must remove the delay jitter in order to faithfully recreate the original data that can be played back at ®xed delay offset [8]. A number of de®nitions for delay jitter have appeared in the literature [10,11]. In this paper, the delay jitter is de®ned as the variation of delay between consecutive packets. If the arrival and departure processes are stationary, the jitter sequence is also stationary. Negative jitters corresponding to a clustering of packets can result in buffer over¯ow. By contrast, positive jitters corresponding to a dispersion of packets can result in excessive delay. For the transmission of voice and video packets over IP networks, delay jitters must be removed using playback mechanism to prevent unacceptable levels of distortion. For example, it has been shown that the typical value of delay jitter is ranging from 70 to 100 ms for transmission of voice packets over frame relay networks [3]. On the other hand, for the transmission of MPEG video over the Internet, the delay jitter can be up to 2 s. In this case, playback buffer is needed for jitter cancellation at receiver end. This paper focuses on the statistics of network delay and delay jitter and their effects on the transmission of voice and video packets over IP networks. Many research works have focused on the analysis of delay and delay jitters. Boyer.et al. [12] proposed an approach using spacing mechanism at network entry point to decrease the `clustering' of periodic traf®c transmission. He presented the effect of periodic background traf®c on the delay and delay jitter performance using simulation. Guilleman and Roberts [13] analyzed the steady-state statistics of delay jitter for periodic traf®c streams that were governed by a Markov structure. Landry and Stavrakakis derived the probability density function (pdf) of delay jitter for a tagged renewal stream multiplexed with un-correlated background traf®c and in®nite buffer. Matragi, Bisdikian, and Sohraby [15] analyzed the pdf of delay jitter for a tagged renewal stream with correlated background traf®c with in®nite buffer system. It calculated the cell inter-departure time distribution for renewal stream on ®rst come ®rst service (FCFS) basis. All of the above studies were performed under the following conditions: the tagged stream must be a renewal or periodic process and the system has buffer capacity. It should be noticed that analysis performed in Refs. [13,14] with the assumption of independent constant
bit-rate (CBR) background traf®c, is not suitable for the high-speed IP networks where the traf®c are usually bursty in nature. This paper focuses on the statistics of network delay and delay jitter due to multiplexing and buffering of voice and video packets at IP level with bursty background traf®c, where encoding/decoding delay and packetization delay of voice or video packets are ignored since voice samples and video frames are packetized into ®xed packets. The statistics of delay and delay jitter focuses on the performance behavior of transient period for tagged traf®c streams when they pass through IP router with realistically ®nite buffer and bursty background traf®c. The analysis is done using queuing transient solution. For a given background traf®c condition, the steady state delay jitter behavior is estimated from the obtained transient statistics when the reference traf®c arrival sequence increases up to large number. The numerical results evaluate the relationship between the delay jitter and the characteristics of the background traf®c, such as burstiness, traf®c load and traf®c burst length, and also their effects on the performance behavior for transmission of voice and video packets over IP networks. The results demonstrate that network delay and delay jitter are the nature of packet-switched networks. Whenever packets are buffered, the information about their inter-packet timing is lost. Buffering is necessary for the contention of resources among different ¯ows but it is able to be minimized using fair-sharing algorithms such as Fair Queuing, Virtual Clock, HRR and PGPS [16±19]. However, the advantage of buffering is to smooth the packet ¯uctuation when allocated bandwidth is less than the peak rate. In this case, buffering is
Fig. 2. Packet delay and delay variation process.
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Fig. 3. PDF of delay at different traf®c load.
determined by traf®c burstiness. Generally speaking, delay jitter increases when traf®c are in bursts.
2. System model In IP networking environment, different kinds of traf®c, such as audio, video and data, go through IP routers across entire networks. Packets are multiplexed in IP routers. There is no jitter if all traf®c streams with the same arrival rate are superposed. In this case, delay is identical for every packet because the traf®c is periodic. Otherwise, as shown in Fig. 1, delay jitter is determined by different arrival rate of traf®c streams. In the following analysis, traf®c streams are assumed to be multiplexed with each other in a single IP router rather than multi-nodes network to investigate the delay jitters characteristics. Considering a typical IP router
interconnected by several different inputs and output links, the router is modeled as non-blocking tandem switching node associated with output queuing system at the output port. The non-blocking switching function includes that IP packet streams carried on each input link are de-multiplexed at the input port and then routed to the corresponding output port according to IP connection table. Even when all IP traf®cs arriving at the input ports are destined to the same output port, the router is fast enough to transfer such IP traf®c to the output port. This assumption enables us to focus on one output port of the router independent of the behavior of the other output ports [6]. Each output port is modeled as a single server queue with ®nite buffer capacity. IP is a connectionless protocol, where each IP PDU of variable length (IP protocol data unit) is treated independently [20]. However, for VoIP and video streaming over IP [22], traf®c are segmented into small and ®xed-size packets
Fig. 4. PDF of delay jitter at different traf®c load.
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Table 1 Delay jitter distribution at different traf®c Load traf®c load
Jitter . 10 ms (%)
Jitter . 20 ms (%)
Jitter . 30 ms (%)
0.77 0.85 0.93
13 28.39 56.5
2.59 10 34.33
0.52 3.5 21
[7] in order to achieve small network delay and delay jitter. This assumption leads to a discrete-time queuing system, in which channel time is slotted with the size being equal to the transmission time of one IP packet. Then, arrival and departure of packets only happen at the slot boundary. Within the same slot, random selection policy is used for processing of arriving IP packets, but for the IP packets arriving at different time slots, the processing is based on FCFS. When queuing buffer is empty, arriving IP packet is transmitted without delay. The following analysis focuses on the performance of the tagged traf®c stream, denoted as T-stream, in which the inter-arrival time between two consecutive IP packets is independent of general distribution probability density, that is f k Probabilityinter-arrival time k slots:
1
The background traf®c, denoted as B-stream, is assumed as the superposition of N independent streams. To represent the bursty nature of traf®c stream, each traf®c stream is modeled as an Interrupt Bernoulli Process (IBP) consisting of a two-state Markov chain, in which the state ON and the state OFF appearing in turns. The transition probability from the state ON to the state OFF is a per slot and from the state OFF to the state ON is b per slot, respectively. Therefore, the duration length of the ON state is geometrically distributed with mean value of 1/a while the duration length of the OFF state is also geometrically distributed with
mean value of 1/b . The fraction of time that the process is in the ON state is b= a 1 b: The superposition of N IBP traf®c streams forms a N 1 1-state Markov chain. Each IBP process generates IP packets at a rate of l per slot only when it is in the ON state. Thus, the average arrival rate of packets from a single IBP process is lb= a 1 b: Let Bik be a random variable representing the number of packets arriving from B-stream at kth time slot with ik IBP processes that are in the ON state. Then the probability density for Bik can be expressed as: 8 ! > i > < k lj 1 2 lik 2j ; 0 # j # ik ; 0 # ik # H bik j 2 j > > : 0; otherwise; where H is the maximum number of the IBP processes. It can be seen that Bik changes slot by slot depending on the values of a and b . In the following sections, the effect of this dynamic behavior of the B-stream on the performance of delay and delay jitter for the T-stream is considered. 3. Performance analysis In order to evaluate the delay and delay jitter for the T-stream, the following analysis is ®rst focusing on the queuing length distribution of the IP router's output queue. As shown in Fig. 2, Qnik ;k is a random variable to represent the queuing length at the end of kth time slot following the
Fig. 5. PDF of delay at different traf®c burstiness.
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Fig. 6. PDF of delay jitter at different traf®c burstiness.
nth packet from the T-stream in the case that ik IBP processes s in the B-stream are in the ON state in the kth slot, where 0 # ik # H (H is the maximum number of the IBP processes). For k 0; Qni0 ;0 is the queuing length immediately before the nth packet arriving from the T-stream. Likewise, Q nik ;k11 is a random variable to represent the number of packets in the queue including the packet being served at the beginning of the k 1 1th time slot following the nth packet arriving from the T-stream where ik IBP processes from the B-stream are in the ON state. At the beginning of the ®rst time slot following the nth packet from the T-stream, there are Q ni0 ;1 packets in the output queue. Q ni0 ;1 includes (1) the number of packets already in the buffer immediately before the nth packet arriving from the T-stream Qni0 ;0 ; (2) one packet newly arriving from the T-stream and (3) Bi0 packets newly arriving from the B-
stream. We can obtain the following equations: Q ni0 ;1 Qni0 ;0 1 1 1 Bi0 ; Q nik ;k11 Qnik ;k 1 Bik ;
0 # i0 # H; k 0; k 1; 2; 3; ¼
3 4
At the beginning of k 1 1th time slot, the number of Bstream in the ON state is changed from ik to ik11 with the probability of ! ! ik X ik H 2 ik i ik 2i bik11 2i 1 1 2 a a pik ;ik11 i i 2 i i0 k11 2 bH2ik 2 ik11 2i ; 0 # i # ik ; 0 # ik11 2 i # H 2 ik ;
Fig. 7. PDF of delay at different traf®c burst length.
5
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Table 2 Delay jitter distribution at different traf®c burstiness Traf®c burstiness
Jitter . 10 ms (%)
Jitter . 20 ms (%)
Jitter . 30 ms (%)
1.25 2.5 5.0
10.5 33.8 52.5
3.6 13.9 31.9
0.7 5.7 20.2
where the term ! ik 1 2 ai a ik 2i i is the probability that i IBP processes from the B-stream remain in the ON state while the other ik 2 i IBP processes change from the ON state to the OFF state at the beginning of k 1 1th slot, and the term ! H 2 ik 1 2 bik11 2i bH2ik 2 ik 2i ik11 2 i represents the probability that ik 2 i IBP processes among H 2 ik change from the OFF state to the ON state. Hence, the relationship between Qnik ;k and Q nik21 ;k can be presented as Qnik ;k
H X ik 2 1 0
pik21 ;ik max Q nik21 ;k 2 1; 0 :
6
qnik ;k j
Let be the probability density of Qnik ;k : Then we have the following equations: qni0 ;1 j qni0 ;0 j 2 1bi0 j 2 1;
7
qnik21 ;k j
8
qnik ;k j
qnik21 ;k21 jbik21 j; H X
ik 2 1 0
pik21 ;ik c qnik21 ;k j 1 1 ;
9
where c g x is a discrete-time operation [21] given by 8 0; x # 0; > > < c g x g 0 1 g 1; x 1; > > : g x; x . 1: When the n 1 1th packet from the T-stream arrives in the kth time slot, the corresponding queuing length is Qnik ;k : Recalling Eq. (1), f(k) is the probability when the inter-arrival time is k time slots. Then the probability density qn11 i0 ;0 j represents the queuing length distribution before the n 1 1th packet arrival from the T-stream and is given by qn11 i0 ;0 j
KX max k0
f kqnik ;k j:
10
It requires an iterative calculation by setting the initial values to q1i0 ;0 j with the condition: H X K X i0
j0
q1i0 ;0 j 1;
11
where K is the buffer length of the output queue. The network delay and delay jitter for the T-stream can be obtained using the queuing length distribution of the output queue. As shown in Fig. 2, when the n 1 1th packet from the Tstream arrives at the kth time slot, the queuing delay can be
Fig. 8. PDF of delay jitter at different traf®c burst length.
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Table 3 Delay jitter distribution at different burst length Burst length
Jitter . 10 ms (%)
Jitter . 20 ms (%)
Jitter . 30 ms (%)
5 slots 10 slots 20 slots
12.51 28.57 46.5
1.95 9.97 26.2
0.3 3.49 15.13
expressed as Qn11 i0 ;0
Qnik ;k
1
Wink ;k ;
Qnik ;k
where packets are already in the queue, and Wink ;k represents the packets from the B-stream arriving at the kth slot together with the n 1 1th packet from the T-stream but are severed before the n 1 1th packet from the T-stream. The waiting time for the n 1 1th packet from the T-stream including (1) the time spent in serving the packets found in the buffer upon the arrival of the n 1 1th packet from the T-stream and (2) the time to serve the packets from the B-stream arriving together with the n 1 1th packet from the T-stream and served before the n 1 1th packet. The random selection policy is used to serve the packets arriving in the same slot. Assuming there are l packets arriving from the B-stream in the kth time slot, then its probability density bik l can be calculated using the Eq. (2). The probability that the n 1 1th packet from the T-stream is jth served among these l 1 1 packets is 1= l 1 1 j # l 1 1: The probability density wnik ;k j for Wink ;k is given by wnik ;k j
ik X lj 2 1
bik l ; l11
two consecutive packets from the T-stream in the output queue. As shown in Fig. 2, the random valuable DJitternik ;k represents the delay jitter between the nth packet and (n 1 1)th packet from the T-stream, then DJitternik ;k can be given by 15 DJitternik ;k Qnik ;k 1 Wink ;k 2 Qni0 ;0 1 Win0 ;0 : Let djitternik ;k j be the probability density for the DJitternik ;k : Then, djitternik ;k is given by djitternik ;k j qnik ;k j 1 q0 wnik ;k j 1 q0 wni0 ;0 2 j 1 q0 ; 16 where q0 Qni0 ;0 is the number of packets in the queue before the nth packet arriving from the T-stream. Through unconditioning Eq. (16), the ®nal delay variation distribution probability density can be expressed as djittern j: djittern j
K X H X q0 0 i0 0
qni0 ;0 q0
8 2 0 139 KX H H
0 # ik # H; 1 # j # K; 12
k
Therefore, the probability density of queuing delay in the output queue for the n 1 1th packet from the T-stream is the convolution of these two probability densities, that is delayn11 j
H X ik 0
qnik ;k jwnik ;k j:
13
17 When n ! 1; the steady-state probability density for delay jitter is Djitter j lim djittern j: n!1
When n ! 1; the steady-state probability density of packet delay in the output queue is given by Delay j lim delayn j:
14
n!1
Delay jitter is de®ned as the variation in delay between the
In each iteration, the convolution operations are required at every time slot and the number of convolution operation is determined by the length of the T-stream inter-arrival time. In this case, FFT should be used to save the computing time and cost.
Table 4 Statistical characteristics of the MPEG VBR-encoded video traces Mean rate (Bits/frame) MovieÐLambs MTV-1 MTV-2 EpisodesÐMrBean SportsÐSoccer TalkÐShow TVÐNews CartoonÐAsterix
761.62 2:7167 £ 104 2:1841 £ 104 1:9485 £ 104 2:7725 £ 104 1:6051 £ 104 1:6957 £ 104 2:4677 £ 104
Maximum rate (Bits/frame) 5
1:4821 £ 10 2:5307 £ 105 2:7760 £ 105 2:5293 £ 105 2:1012 £ 105 1:1789 £ 105 2:0967 £ 105 1:6273 £ 105
Minimum rate (Bits/frame) 318.00 406.32 530.00 379.81 3259.5 2296.68 300.32 335.69
Standard Div. (Bits/frame) 4
1:2362 £ 10 2:5465 £ 104 2:3688 £ 104 2:2784 £ 104 2:3475 £ 104 1:8240 £ 104 2:1537 £ 104 2:2242 £ 104
Cov. (Bits/frame) 3:6041 £ 105 1:5294 £ 106 1:3233 £ 106 1:2253 £ 106 1:2997 £ 106 7:8469 £ 105 1:0940 £ 106 1:1668 £ 106
870
L. Zhang et al. / Computer Communications 25 (2002) 863±873
Fig. 9. The playback of MPEG video trace `Cantoon_Asterix'.
4. Numerical results and discussion The following numerical results focus on the effects of delay jitter in terms of pdf on the performance of the tagged voice IP stream under different traf®c conditions. The voice IP stream is obtained from a speech sample sequence in which the talk-spurt period is encoded at a constant bit rate of 64 Kbits/s and the silence period of the speech is when no data is generated. The voice speech is packetized into ®xed IP packets of 128 bytes and then is transmitted over an E1 line (2.048 Mbits/s). The equivalent transmission time for each voice IP packet is 0.5 ms. The interarrival time between two consecutive voice packets is 10 ms which is equivalent to 20 time slots. The receiving buffer is 150 packets. Figs. 3 and 4 show the pdf of delay and delay jitter for the tagged voice packet stream under different traf®c loading conditions. It can be seen that the increase of traf®c load makes the distribution of jitter pdf to have longer tails across
wide range (corresponding to large jitters) and lower peak values (corresponding to less probability for zero jitter). For example, as shown in Table 1, when the traf®c load is 0.77, 97% of delay jitters range from 220 to 20 ms, by contrast, if the traf®c load increases to 0.93, the percentage of delay jitters ranging from 220 to 20 ms is reduced to 65%, which means that the delay jitters have more effects on the performance. Figs. 5 and 6 present the pdf of delay and delay jitter for the T-stream with different bursty background traf®c conditions. It can be seen that when traf®c burstiness increases, the jitter performance becomes worse (Fig. 7). For example, as shown in Table 2, for traf®c burstiness of 1.25, 17% of packets have the zero jitter, and 99.8% of delay jitters range from 220 to 20 ms. By contrast, for traf®c burstiness of 5.0, only 7% of packets have the zero jitter and 32% of delay jitters exceed ^20 ms.Therefore, delay jitters are sensitive to traf®c burstiness which may seriously affect the service quality for VoIP.
Fig. 10. Video packetization format.
L. Zhang et al. / Computer Communications 25 (2002) 863±873
871
Fig. 11. PDF of frame delay jitter at different video streams.
Fig. 8 presents the pdf of delay jitter for the T-stream with different burst lengths, where traf®c load and traf®c burstiness are kept as constant. It can be seen that the bursty background traf®c with longer burst-length makes the distribution of jitter to have longer tails across wide range (corresponding to large jitters) and lower peak values (corresponding to less probability for zero jitter). For example, as shown in Table 3 and Fig. 8, when the burst-length of each background traf®c stream is ®ve slots, 15% of packets have the zero jitter and 98% of delay jitters range from 220 to 20 ms. By contrast, for the burst-length of 20 slots, only 5% of packets have the zero jitter and 74% of delay jitters range from 220 to 20 ms. Therefore, the burst-length of background traf®c stream affects the jitter performance signi®cantly. The effects of delay jitter on the performance of MPEG video streams over the IP are also demonstrated. For the illustrative purpose only, real-time MPEG video traces with different contents including Movies, Sports, News, Talk Shows, MTVs and Cartoons are used in the evaluation. These MPEG video traces are decoded and extracted into individual frames using a public-domain software Berkeley MPEG version 1.3 [1,4]. The statistical characteristics of these video traces are shown in Table 4. Fig. 9 shows the playback of MPEG Video trace `Cantoon_Asterix'.
These MPEG video traces were encoded with the following parameters: Encoder input: 384 £ 288 pel. Color format: YUV (4:1:1, resolution of 8 bits). Quantization value: I 10; P 14; B 18: Pattern: IBBPBBPBBPBB. GOP size: 12. Motion vector search: `Logarithmic'/`Simple'. Reference frame: `Original'. Slices: 1. Vector/range: half pel/10. Frame rate: 25 frames/s. Video length: 40,000 frames. As shown in Fig. 10, since video bit stream is packetized into ®xed-length IP packets and the number of bits contained in video frames are varied from frame to frame, after packetization, the number of video packets contained in each frame is also varied from frame to frame. The video packets contained in each video frame is transmitted using ®xed inter-arrival time but it is varied from frame to frame. Therefore, the frame delay jitter is de®ned as the delay jitter between two consecutive frames. It can be estimated from the delay jitter between the last packets of the two consecutive frames.
Table 5 Statistical characteristics of video streams MPEG video traces
Maximum rate
Minimum rate
Mean rate
Standard deviation
Maximum rate/Mean rate
SportsÐSoccer CartoonÐAsterix TalkÐShow
13.0427 9.0493 8.8973
0.3540 0.0373 0.3487
2.0432 1.7006 2.0030
2.2692 1.7762 1.7158
6.3835 5.3212 4.4420
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Table 6 Delay jitter distribution of different video streams MPEG video traces
Jitter . 20 ms (%)
Jitter . 40 ms (%)
Jitter . 60 ms (%)
TalkÐShow CartoonÐAsterix SportsÐSoccer
33.37 41.91 53.03
13.64 19.07 26.95
5.79 12.02 15.04
Fig. 11 presents the pdf of delay jitter for three different MPEG video streams with the same background traf®c conditions where overall traf®c load is 0.88. The statistical characteristics of these video traces are shown in Table 5, where Max. stands for the maximum rate or peak rate, Min. for the minimum rate, Mean for the mean rate and Std. for the standard deviation. From Fig. 11 and Table 5, it can be seen that since the video traf®c stream is more bursty in nature compared to the voice traf®c streams, the video streams have more signi®cant effects on the performance of delay jitters than the voice traf®c streams have as shown in Fig. 8. On the other hand, it can also be seen that the burst video traf®c corresponds to signi®cant delay jitter distribution that has longer tails across wide range (corresponding to large jitters) and lower peak values (corresponding to less probability for zero jitter). For example, as shown in Table 6, since the sports video stream with Std. of 2.2692 and Max./Mean of 6.3835 is more bursty than that of TalkÐShow video stream with Std. of 1.7158 and Max./Mean of 4.4420, it has 26.95% of frame delay jitter over more than 40 ms. By contrast, the TalkÐShow video stream only has 13.64% of frame delay jitter more than 40 ms. The MPEG includes a series of audio±visual standard known as MPEG-1 and MPEG-2 [5] which are the ®rst international standards in the ®eld of high-quality digital
audio compression. The following numerical results of inter-stream delay jitter between MPEG video and audio streams are evaluated using a CartoonÐAsterix as the tagged video stream and a 24 ms 1152 PCM audio clip with a sample rate of 48 kHz and encoded at 16 bits per sample. Fig. 12 presents the pdf of inter-streams delay jitter under the condition of different background traf®c burstiness. It can be seen that large background traf®c burstiness has more signi®cant effects on the distribution of inter-streams delay jitter with longer tails across wide range (corresponding to large jitters) and lower peak values (corresponding to less probability for zero jitter). For example, as shown in Table 7, when the background traf®c burstiness is 1.25, 21.49% of inter-streams delay jitters are ranging from 240 to 40 ms. On the other hand, when the background traf®c burstiness increases to 2.5, correspondingly, 51.05% of inter-streams delay jitters exceed 40 ms. Then the service quality is deteriorated seriously. 5. Conclusion Network delay and delay jitter are the important QoS parameters for real-time services, such as packet voice and packet video traf®c over the Internet. The delay jitter
Fig. 12. PDF of frame delay jitter between video and audio at different traf®c burstiness.
L. Zhang et al. / Computer Communications 25 (2002) 863±873 Table 7 Delay jitter distribution between video and audio at different burstiness Burstiness Jitter . 20 ms (%) Jitter . 40 ms (%) Jitter . 60 ms (%) 1.25 1.875 2.5
33.33 48.68 64.33
21.49 24.44 51.05
10.02 20.89 38.07
can be either positive or negative. A sequence of negative jitter (clustering) may result in downstream nodal congestion and consecutive packet loss. On the other hand, a sequence of positive jitters (dispersion) may result in consecutive packet signi®cant delays. The pdf of delay jitter provides high-order distribution of jitter length that demonstrates the jitter behavior in high-speed networks. Since the estimation of delay jitters is a dif®cult task due to its transient queuing effects, this paper focuses on the performance behavior of delay and delay jitter by taking into account of the dynamic nature of the background traf®c using transient queuing solution. The steady state solution of delay jitter is archived when the sequence of the tagged stream arrival goes to a large number. The obtained numerical results demonstrate the effects of delay and delay jitter on speech clips, video traces and interstreams between video and audio streams. It shows that network conditions including traf®c load, traf®c burstiness and burst-length all have signi®cant effects on the pdf of delay and delay jitters. It can be seen that the inter-streams delay jitter may in¯uence the service quality. On the other hand, large sequence of either clustered or dispersed packets have signi®cant effects on the performance of QoS. Since the effect of delay jitter on MPEG video stream can be removed using the playback buffer at the receiver end, the network engineers may need to pay more attentions to the effect of delay jitter on voice speech. References [1] MPEG traces available via anonymous FTP from ftp://ftp-info3. informatik.uni uerzburg.de/pub/MPEG/. [2] Telogy Networks, Voice over IP (VoIP), http://www.telogy.com. [3] Telogy Networks, Voice over Packet, White paper, http://www. telogy.com.
873
[4] O. Rose, Statistical properties of MPEG video traf®c and their impact on traf®c modeling in ATM systems, Proceedings of the 20th Annual Conference on Local Computer Networks, 1995, pp. 397±406. [5] P. Noll, MPEG digital audio coding, IEEE Signal Process. Mag. 14 (5) (1997) 59±81. [6] M. Wernik, O. Aboul-Magd, H. Gilbert, Traf®c management for BISDN services, IEEE Network 6 (5) (1992) 10±19. [7] C.-N. Chuah, R. H. Katz, Network provisioning and resource management for ip telephony, University of California, Berkeley, Report No. UCB/CSD-99-1061, September 1, 1999. [8] Z. Wang, J. Crowcroft, Analysis of burstiness and jitter in real-time communications, SIGCOMM'93, 1993, pp. 13±19. [9] ITU-T Recommendation G.114, General characteristics of international telephone connections and international telephone circuits: one-way transmission time, February, 1996. [10] J. Roberts, F. Guillemin, Jitter in ATM networks and its impact on peak rate enforcement, Perf. Eval. 16 (1992) 35±48. [11] F. Guillemin, W. Monin, Management of cell delay variation in ATM networks, Proceedings of the IEEE GLOBECOM'92, 1 (1992) pp. 128±132. [12] P. Boyer, F. Guillemin, M. Servel, J.P. Coudreuse, Spacing cells protects and enhances utilization of ATM network links, IEEE Network 6 (5) (1992) 38±49. [13] F. Guilleman, J.W. Roberts, Jitter and bandwidth enforcement, Proceedings of the IEEE Globecom'91, 1 (1991) pp. 261±265. [14] R. Landry, I. Stavrakakis, Traf®c shaping of a tagged stream in an ATM network: approximate end-to-end analysis, Proceedings of the IEEE Infocom'95, 1 (1995) pp. 162±169. [15] W. Matragi, C. Bisdikian, K. Sohraby, Jitter calculus in ATM networks: single node case, Proceedings of the IEEE Infocom'94, 1 (1994) pp. 232±241. [16] A. Demers, S. Keshav, S. Shenker, Analysis and simulation of a fair queueing algorithm, Proceedings of ACM SIGCOMM'89, 1989, pp. 1±12. [17] C. Kalmanek, H. Kanakia, S. Keshav, Rate controlled servers for very high-speed networks, Proceedings of IEEE GlobeCom'90, 1 (1990) pp. 12±20. [18] L. Zhang, Virtual clock: a new traf®c control algorithm for packet switching networks, Proceedings of ACM SIGCOMM'90, 1990, pp. 19±29. [19] A. Parekh, R. Gallager, A generalized processor sharing approach to ¯ow controlÐthe single node case, Proceedings of IEEE INFOCOM'92, 2 (1992) pp. 915±924. [20] J. Postel, Internet protocol, RFC-791, September, 1981. [21] P. Tran-Gia, Discrete-time analysis for inter- departure distribution of GI/G/1 queues, Proceedings of Teletraf®c Analysis and Computer Performance Evaluation, North Holland, Elsevier Science, Amsterdam, 1986. [22] K. Sklower, B. Lloyd, D. Carr, The PPP multilink protocol (MP), RFC 1717.
www.elsevier.com/locate/comcom
Effect of delay and delay jitter on voice/video over IP Liren Zhang*, Li Zheng, Koh Soo Ngee Network Technology Research Center, School of Electrical and Electrical Engineering, Nanyang Technological University, Singapore, Singapore 639798 Received 28 August 2001; accepted 3 September 2001
Abstract Internet, especially in the situation when network has bursty background traf®c. However, it is usually dif®cult to evaluate the statistics of delay and delay jitter in such bursty traf®c environment due to their transient queuing nature. This paper focuses on the performance behavior of delay and delay jitter at IP packet level and also their effects on the performance of voice and video packet streams when they are multiplexed with bursty background traf®c in IP routers. The performance analysis of delay and delay jitter is done using the transient queuing solution technique. The numerical results are presented in terms of probability density functions (pdfs) of delay and delay jitter, which focus on their effects on speech samples and video frames. q 2002 Elsevier Science B.V. All rights reserved. Keywords: Voice over IP; Video over IP; QoS; Delay; Delay jitter
1. Introduction In today's networking, transmission of voice and video over the Internet are attractive alternatives both to conventional public telephony and leased lines. Voice over IP (VoIP) refers to real-time delivery of voice packets across networks using the Internet protocol. An appeal of VoIP is based on the capability to facilitate voice and data convergence at application layer. However, this integration effectively ®lls up the data communication channels and ef®ciently provides bandwidth consolidation. It is more important that the Public Switched Telephone Networks (PSTN) toll services can be bypassed using the Internet backbone to slash in prices for long distance calls [2]. On the other hand, video over the Internet has also being seen as the ideal last-mile solution for cable modem, DSL, and wireless networks since it allows service providers to bundle their offerings. Voice and video services often have more restrictive QoS requirement on delay and delay jitter than data transfer applications have, one of the typical problems with the transmission of voice packets and video packets over IP is the QoS guarantee. In fact, the QoS for such video or voice services can be affected by delay, delay jitter and unreliable packet delivery, which are the typical characteristics of the basic IP-network service, because packets are naturally transported using the best effort mechanism on ®rst-come, * Corresponding author. Fax: 161-792-0415. E-mail address: [email protected] (L. Zhang).
®rst-served (FIFO) basis. On the other hand, the size of data packet is variable in nature that allows large data ®le transfers to take the advantage of ef®ciency associated with larger packet sizes. However, these characteristics may contribute large delays and large delay jitters to packet streams, which are the most important concerns of the QoS for voice and video over the Internet. If the network is not properly controlled to meet the QoS requirements, the quality of voice and video is certainly affected. This is particularly true when voice or video packets are transmitted over the public Internet, where voice or video users have few options to secure the end-to-end QoS in terms of delay and delay jitters. ITU-T Recommendation G.114 [9] speci®es that one-way transmission time for connections with adequately controlled echo of voice should be ranging from 0 to 150 ms to be acceptable for most user applications [7]. However, video streaming over IP is able to handle a large jitter margin up to 2 s by using playback buffer at the receiver end. For an end-to-end VoIP call connection, three types of delay are experienced [3]: accumulation delay, processing delay and network delay. The accumulation delay is ranging from 125 ms to several milliseconds that are caused by the processing of voice samples. The processing delay is caused by the packetization of voice samples. The network delay is caused by multiplexing and buffering of voice packets when they are transmitted across networks. Delay jitter is another important parameter that should be taken into account in the support of voice and video services over IP networks. When voice or video packets are
0140-3664/02/$ - see front matter q 2002 Elsevier Science B.V. All rights reserved. PII: S 0140-366 4(01)00418-2
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Fig. 1. Queuing model for delay jitter.
transmitted over the Internet, packets may experience variable delay, called delay jitter. For non-real-time data transfer applications, such as ftp and telnet, delay jitter has little real impact on QoS. By contrast, real-time applications such as audio, video and voice must remove the delay jitter in order to faithfully recreate the original data that can be played back at ®xed delay offset [8]. A number of de®nitions for delay jitter have appeared in the literature [10,11]. In this paper, the delay jitter is de®ned as the variation of delay between consecutive packets. If the arrival and departure processes are stationary, the jitter sequence is also stationary. Negative jitters corresponding to a clustering of packets can result in buffer over¯ow. By contrast, positive jitters corresponding to a dispersion of packets can result in excessive delay. For the transmission of voice and video packets over IP networks, delay jitters must be removed using playback mechanism to prevent unacceptable levels of distortion. For example, it has been shown that the typical value of delay jitter is ranging from 70 to 100 ms for transmission of voice packets over frame relay networks [3]. On the other hand, for the transmission of MPEG video over the Internet, the delay jitter can be up to 2 s. In this case, playback buffer is needed for jitter cancellation at receiver end. This paper focuses on the statistics of network delay and delay jitter and their effects on the transmission of voice and video packets over IP networks. Many research works have focused on the analysis of delay and delay jitters. Boyer.et al. [12] proposed an approach using spacing mechanism at network entry point to decrease the `clustering' of periodic traf®c transmission. He presented the effect of periodic background traf®c on the delay and delay jitter performance using simulation. Guilleman and Roberts [13] analyzed the steady-state statistics of delay jitter for periodic traf®c streams that were governed by a Markov structure. Landry and Stavrakakis derived the probability density function (pdf) of delay jitter for a tagged renewal stream multiplexed with un-correlated background traf®c and in®nite buffer. Matragi, Bisdikian, and Sohraby [15] analyzed the pdf of delay jitter for a tagged renewal stream with correlated background traf®c with in®nite buffer system. It calculated the cell inter-departure time distribution for renewal stream on ®rst come ®rst service (FCFS) basis. All of the above studies were performed under the following conditions: the tagged stream must be a renewal or periodic process and the system has buffer capacity. It should be noticed that analysis performed in Refs. [13,14] with the assumption of independent constant
bit-rate (CBR) background traf®c, is not suitable for the high-speed IP networks where the traf®c are usually bursty in nature. This paper focuses on the statistics of network delay and delay jitter due to multiplexing and buffering of voice and video packets at IP level with bursty background traf®c, where encoding/decoding delay and packetization delay of voice or video packets are ignored since voice samples and video frames are packetized into ®xed packets. The statistics of delay and delay jitter focuses on the performance behavior of transient period for tagged traf®c streams when they pass through IP router with realistically ®nite buffer and bursty background traf®c. The analysis is done using queuing transient solution. For a given background traf®c condition, the steady state delay jitter behavior is estimated from the obtained transient statistics when the reference traf®c arrival sequence increases up to large number. The numerical results evaluate the relationship between the delay jitter and the characteristics of the background traf®c, such as burstiness, traf®c load and traf®c burst length, and also their effects on the performance behavior for transmission of voice and video packets over IP networks. The results demonstrate that network delay and delay jitter are the nature of packet-switched networks. Whenever packets are buffered, the information about their inter-packet timing is lost. Buffering is necessary for the contention of resources among different ¯ows but it is able to be minimized using fair-sharing algorithms such as Fair Queuing, Virtual Clock, HRR and PGPS [16±19]. However, the advantage of buffering is to smooth the packet ¯uctuation when allocated bandwidth is less than the peak rate. In this case, buffering is
Fig. 2. Packet delay and delay variation process.
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Fig. 3. PDF of delay at different traf®c load.
determined by traf®c burstiness. Generally speaking, delay jitter increases when traf®c are in bursts.
2. System model In IP networking environment, different kinds of traf®c, such as audio, video and data, go through IP routers across entire networks. Packets are multiplexed in IP routers. There is no jitter if all traf®c streams with the same arrival rate are superposed. In this case, delay is identical for every packet because the traf®c is periodic. Otherwise, as shown in Fig. 1, delay jitter is determined by different arrival rate of traf®c streams. In the following analysis, traf®c streams are assumed to be multiplexed with each other in a single IP router rather than multi-nodes network to investigate the delay jitters characteristics. Considering a typical IP router
interconnected by several different inputs and output links, the router is modeled as non-blocking tandem switching node associated with output queuing system at the output port. The non-blocking switching function includes that IP packet streams carried on each input link are de-multiplexed at the input port and then routed to the corresponding output port according to IP connection table. Even when all IP traf®cs arriving at the input ports are destined to the same output port, the router is fast enough to transfer such IP traf®c to the output port. This assumption enables us to focus on one output port of the router independent of the behavior of the other output ports [6]. Each output port is modeled as a single server queue with ®nite buffer capacity. IP is a connectionless protocol, where each IP PDU of variable length (IP protocol data unit) is treated independently [20]. However, for VoIP and video streaming over IP [22], traf®c are segmented into small and ®xed-size packets
Fig. 4. PDF of delay jitter at different traf®c load.
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Table 1 Delay jitter distribution at different traf®c Load traf®c load
Jitter . 10 ms (%)
Jitter . 20 ms (%)
Jitter . 30 ms (%)
0.77 0.85 0.93
13 28.39 56.5
2.59 10 34.33
0.52 3.5 21
[7] in order to achieve small network delay and delay jitter. This assumption leads to a discrete-time queuing system, in which channel time is slotted with the size being equal to the transmission time of one IP packet. Then, arrival and departure of packets only happen at the slot boundary. Within the same slot, random selection policy is used for processing of arriving IP packets, but for the IP packets arriving at different time slots, the processing is based on FCFS. When queuing buffer is empty, arriving IP packet is transmitted without delay. The following analysis focuses on the performance of the tagged traf®c stream, denoted as T-stream, in which the inter-arrival time between two consecutive IP packets is independent of general distribution probability density, that is f k Probabilityinter-arrival time k slots:
1
The background traf®c, denoted as B-stream, is assumed as the superposition of N independent streams. To represent the bursty nature of traf®c stream, each traf®c stream is modeled as an Interrupt Bernoulli Process (IBP) consisting of a two-state Markov chain, in which the state ON and the state OFF appearing in turns. The transition probability from the state ON to the state OFF is a per slot and from the state OFF to the state ON is b per slot, respectively. Therefore, the duration length of the ON state is geometrically distributed with mean value of 1/a while the duration length of the OFF state is also geometrically distributed with
mean value of 1/b . The fraction of time that the process is in the ON state is b= a 1 b: The superposition of N IBP traf®c streams forms a N 1 1-state Markov chain. Each IBP process generates IP packets at a rate of l per slot only when it is in the ON state. Thus, the average arrival rate of packets from a single IBP process is lb= a 1 b: Let Bik be a random variable representing the number of packets arriving from B-stream at kth time slot with ik IBP processes that are in the ON state. Then the probability density for Bik can be expressed as: 8 ! > i > < k lj 1 2 lik 2j ; 0 # j # ik ; 0 # ik # H bik j 2 j > > : 0; otherwise; where H is the maximum number of the IBP processes. It can be seen that Bik changes slot by slot depending on the values of a and b . In the following sections, the effect of this dynamic behavior of the B-stream on the performance of delay and delay jitter for the T-stream is considered. 3. Performance analysis In order to evaluate the delay and delay jitter for the T-stream, the following analysis is ®rst focusing on the queuing length distribution of the IP router's output queue. As shown in Fig. 2, Qnik ;k is a random variable to represent the queuing length at the end of kth time slot following the
Fig. 5. PDF of delay at different traf®c burstiness.
L. Zhang et al. / Computer Communications 25 (2002) 863±873
867
Fig. 6. PDF of delay jitter at different traf®c burstiness.
nth packet from the T-stream in the case that ik IBP processes s in the B-stream are in the ON state in the kth slot, where 0 # ik # H (H is the maximum number of the IBP processes). For k 0; Qni0 ;0 is the queuing length immediately before the nth packet arriving from the T-stream. Likewise, Q nik ;k11 is a random variable to represent the number of packets in the queue including the packet being served at the beginning of the k 1 1th time slot following the nth packet arriving from the T-stream where ik IBP processes from the B-stream are in the ON state. At the beginning of the ®rst time slot following the nth packet from the T-stream, there are Q ni0 ;1 packets in the output queue. Q ni0 ;1 includes (1) the number of packets already in the buffer immediately before the nth packet arriving from the T-stream Qni0 ;0 ; (2) one packet newly arriving from the T-stream and (3) Bi0 packets newly arriving from the B-
stream. We can obtain the following equations: Q ni0 ;1 Qni0 ;0 1 1 1 Bi0 ; Q nik ;k11 Qnik ;k 1 Bik ;
0 # i0 # H; k 0; k 1; 2; 3; ¼
3 4
At the beginning of k 1 1th time slot, the number of Bstream in the ON state is changed from ik to ik11 with the probability of ! ! ik X ik H 2 ik i ik 2i bik11 2i 1 1 2 a a pik ;ik11 i i 2 i i0 k11 2 bH2ik 2 ik11 2i ; 0 # i # ik ; 0 # ik11 2 i # H 2 ik ;
Fig. 7. PDF of delay at different traf®c burst length.
5
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Table 2 Delay jitter distribution at different traf®c burstiness Traf®c burstiness
Jitter . 10 ms (%)
Jitter . 20 ms (%)
Jitter . 30 ms (%)
1.25 2.5 5.0
10.5 33.8 52.5
3.6 13.9 31.9
0.7 5.7 20.2
where the term ! ik 1 2 ai a ik 2i i is the probability that i IBP processes from the B-stream remain in the ON state while the other ik 2 i IBP processes change from the ON state to the OFF state at the beginning of k 1 1th slot, and the term ! H 2 ik 1 2 bik11 2i bH2ik 2 ik 2i ik11 2 i represents the probability that ik 2 i IBP processes among H 2 ik change from the OFF state to the ON state. Hence, the relationship between Qnik ;k and Q nik21 ;k can be presented as Qnik ;k
H X ik 2 1 0
pik21 ;ik max Q nik21 ;k 2 1; 0 :
6
qnik ;k j
Let be the probability density of Qnik ;k : Then we have the following equations: qni0 ;1 j qni0 ;0 j 2 1bi0 j 2 1;
7
qnik21 ;k j
8
qnik ;k j
qnik21 ;k21 jbik21 j; H X
ik 2 1 0
pik21 ;ik c qnik21 ;k j 1 1 ;
9
where c g x is a discrete-time operation [21] given by 8 0; x # 0; > > < c g x g 0 1 g 1; x 1; > > : g x; x . 1: When the n 1 1th packet from the T-stream arrives in the kth time slot, the corresponding queuing length is Qnik ;k : Recalling Eq. (1), f(k) is the probability when the inter-arrival time is k time slots. Then the probability density qn11 i0 ;0 j represents the queuing length distribution before the n 1 1th packet arrival from the T-stream and is given by qn11 i0 ;0 j
KX max k0
f kqnik ;k j:
10
It requires an iterative calculation by setting the initial values to q1i0 ;0 j with the condition: H X K X i0
j0
q1i0 ;0 j 1;
11
where K is the buffer length of the output queue. The network delay and delay jitter for the T-stream can be obtained using the queuing length distribution of the output queue. As shown in Fig. 2, when the n 1 1th packet from the Tstream arrives at the kth time slot, the queuing delay can be
Fig. 8. PDF of delay jitter at different traf®c burst length.
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Table 3 Delay jitter distribution at different burst length Burst length
Jitter . 10 ms (%)
Jitter . 20 ms (%)
Jitter . 30 ms (%)
5 slots 10 slots 20 slots
12.51 28.57 46.5
1.95 9.97 26.2
0.3 3.49 15.13
expressed as Qn11 i0 ;0
Qnik ;k
1
Wink ;k ;
Qnik ;k
where packets are already in the queue, and Wink ;k represents the packets from the B-stream arriving at the kth slot together with the n 1 1th packet from the T-stream but are severed before the n 1 1th packet from the T-stream. The waiting time for the n 1 1th packet from the T-stream including (1) the time spent in serving the packets found in the buffer upon the arrival of the n 1 1th packet from the T-stream and (2) the time to serve the packets from the B-stream arriving together with the n 1 1th packet from the T-stream and served before the n 1 1th packet. The random selection policy is used to serve the packets arriving in the same slot. Assuming there are l packets arriving from the B-stream in the kth time slot, then its probability density bik l can be calculated using the Eq. (2). The probability that the n 1 1th packet from the T-stream is jth served among these l 1 1 packets is 1= l 1 1 j # l 1 1: The probability density wnik ;k j for Wink ;k is given by wnik ;k j
ik X lj 2 1
bik l ; l11
two consecutive packets from the T-stream in the output queue. As shown in Fig. 2, the random valuable DJitternik ;k represents the delay jitter between the nth packet and (n 1 1)th packet from the T-stream, then DJitternik ;k can be given by 15 DJitternik ;k Qnik ;k 1 Wink ;k 2 Qni0 ;0 1 Win0 ;0 : Let djitternik ;k j be the probability density for the DJitternik ;k : Then, djitternik ;k is given by djitternik ;k j qnik ;k j 1 q0 wnik ;k j 1 q0 wni0 ;0 2 j 1 q0 ; 16 where q0 Qni0 ;0 is the number of packets in the queue before the nth packet arriving from the T-stream. Through unconditioning Eq. (16), the ®nal delay variation distribution probability density can be expressed as djittern j: djittern j
K X H X q0 0 i0 0
qni0 ;0 q0
8 2 0 139 KX H H
0 # ik # H; 1 # j # K; 12
k
Therefore, the probability density of queuing delay in the output queue for the n 1 1th packet from the T-stream is the convolution of these two probability densities, that is delayn11 j
H X ik 0
qnik ;k jwnik ;k j:
13
17 When n ! 1; the steady-state probability density for delay jitter is Djitter j lim djittern j: n!1
When n ! 1; the steady-state probability density of packet delay in the output queue is given by Delay j lim delayn j:
14
n!1
Delay jitter is de®ned as the variation in delay between the
In each iteration, the convolution operations are required at every time slot and the number of convolution operation is determined by the length of the T-stream inter-arrival time. In this case, FFT should be used to save the computing time and cost.
Table 4 Statistical characteristics of the MPEG VBR-encoded video traces Mean rate (Bits/frame) MovieÐLambs MTV-1 MTV-2 EpisodesÐMrBean SportsÐSoccer TalkÐShow TVÐNews CartoonÐAsterix
761.62 2:7167 £ 104 2:1841 £ 104 1:9485 £ 104 2:7725 £ 104 1:6051 £ 104 1:6957 £ 104 2:4677 £ 104
Maximum rate (Bits/frame) 5
1:4821 £ 10 2:5307 £ 105 2:7760 £ 105 2:5293 £ 105 2:1012 £ 105 1:1789 £ 105 2:0967 £ 105 1:6273 £ 105
Minimum rate (Bits/frame) 318.00 406.32 530.00 379.81 3259.5 2296.68 300.32 335.69
Standard Div. (Bits/frame) 4
1:2362 £ 10 2:5465 £ 104 2:3688 £ 104 2:2784 £ 104 2:3475 £ 104 1:8240 £ 104 2:1537 £ 104 2:2242 £ 104
Cov. (Bits/frame) 3:6041 £ 105 1:5294 £ 106 1:3233 £ 106 1:2253 £ 106 1:2997 £ 106 7:8469 £ 105 1:0940 £ 106 1:1668 £ 106
870
L. Zhang et al. / Computer Communications 25 (2002) 863±873
Fig. 9. The playback of MPEG video trace `Cantoon_Asterix'.
4. Numerical results and discussion The following numerical results focus on the effects of delay jitter in terms of pdf on the performance of the tagged voice IP stream under different traf®c conditions. The voice IP stream is obtained from a speech sample sequence in which the talk-spurt period is encoded at a constant bit rate of 64 Kbits/s and the silence period of the speech is when no data is generated. The voice speech is packetized into ®xed IP packets of 128 bytes and then is transmitted over an E1 line (2.048 Mbits/s). The equivalent transmission time for each voice IP packet is 0.5 ms. The interarrival time between two consecutive voice packets is 10 ms which is equivalent to 20 time slots. The receiving buffer is 150 packets. Figs. 3 and 4 show the pdf of delay and delay jitter for the tagged voice packet stream under different traf®c loading conditions. It can be seen that the increase of traf®c load makes the distribution of jitter pdf to have longer tails across
wide range (corresponding to large jitters) and lower peak values (corresponding to less probability for zero jitter). For example, as shown in Table 1, when the traf®c load is 0.77, 97% of delay jitters range from 220 to 20 ms, by contrast, if the traf®c load increases to 0.93, the percentage of delay jitters ranging from 220 to 20 ms is reduced to 65%, which means that the delay jitters have more effects on the performance. Figs. 5 and 6 present the pdf of delay and delay jitter for the T-stream with different bursty background traf®c conditions. It can be seen that when traf®c burstiness increases, the jitter performance becomes worse (Fig. 7). For example, as shown in Table 2, for traf®c burstiness of 1.25, 17% of packets have the zero jitter, and 99.8% of delay jitters range from 220 to 20 ms. By contrast, for traf®c burstiness of 5.0, only 7% of packets have the zero jitter and 32% of delay jitters exceed ^20 ms.Therefore, delay jitters are sensitive to traf®c burstiness which may seriously affect the service quality for VoIP.
Fig. 10. Video packetization format.
L. Zhang et al. / Computer Communications 25 (2002) 863±873
871
Fig. 11. PDF of frame delay jitter at different video streams.
Fig. 8 presents the pdf of delay jitter for the T-stream with different burst lengths, where traf®c load and traf®c burstiness are kept as constant. It can be seen that the bursty background traf®c with longer burst-length makes the distribution of jitter to have longer tails across wide range (corresponding to large jitters) and lower peak values (corresponding to less probability for zero jitter). For example, as shown in Table 3 and Fig. 8, when the burst-length of each background traf®c stream is ®ve slots, 15% of packets have the zero jitter and 98% of delay jitters range from 220 to 20 ms. By contrast, for the burst-length of 20 slots, only 5% of packets have the zero jitter and 74% of delay jitters range from 220 to 20 ms. Therefore, the burst-length of background traf®c stream affects the jitter performance signi®cantly. The effects of delay jitter on the performance of MPEG video streams over the IP are also demonstrated. For the illustrative purpose only, real-time MPEG video traces with different contents including Movies, Sports, News, Talk Shows, MTVs and Cartoons are used in the evaluation. These MPEG video traces are decoded and extracted into individual frames using a public-domain software Berkeley MPEG version 1.3 [1,4]. The statistical characteristics of these video traces are shown in Table 4. Fig. 9 shows the playback of MPEG Video trace `Cantoon_Asterix'.
These MPEG video traces were encoded with the following parameters: Encoder input: 384 £ 288 pel. Color format: YUV (4:1:1, resolution of 8 bits). Quantization value: I 10; P 14; B 18: Pattern: IBBPBBPBBPBB. GOP size: 12. Motion vector search: `Logarithmic'/`Simple'. Reference frame: `Original'. Slices: 1. Vector/range: half pel/10. Frame rate: 25 frames/s. Video length: 40,000 frames. As shown in Fig. 10, since video bit stream is packetized into ®xed-length IP packets and the number of bits contained in video frames are varied from frame to frame, after packetization, the number of video packets contained in each frame is also varied from frame to frame. The video packets contained in each video frame is transmitted using ®xed inter-arrival time but it is varied from frame to frame. Therefore, the frame delay jitter is de®ned as the delay jitter between two consecutive frames. It can be estimated from the delay jitter between the last packets of the two consecutive frames.
Table 5 Statistical characteristics of video streams MPEG video traces
Maximum rate
Minimum rate
Mean rate
Standard deviation
Maximum rate/Mean rate
SportsÐSoccer CartoonÐAsterix TalkÐShow
13.0427 9.0493 8.8973
0.3540 0.0373 0.3487
2.0432 1.7006 2.0030
2.2692 1.7762 1.7158
6.3835 5.3212 4.4420
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Table 6 Delay jitter distribution of different video streams MPEG video traces
Jitter . 20 ms (%)
Jitter . 40 ms (%)
Jitter . 60 ms (%)
TalkÐShow CartoonÐAsterix SportsÐSoccer
33.37 41.91 53.03
13.64 19.07 26.95
5.79 12.02 15.04
Fig. 11 presents the pdf of delay jitter for three different MPEG video streams with the same background traf®c conditions where overall traf®c load is 0.88. The statistical characteristics of these video traces are shown in Table 5, where Max. stands for the maximum rate or peak rate, Min. for the minimum rate, Mean for the mean rate and Std. for the standard deviation. From Fig. 11 and Table 5, it can be seen that since the video traf®c stream is more bursty in nature compared to the voice traf®c streams, the video streams have more signi®cant effects on the performance of delay jitters than the voice traf®c streams have as shown in Fig. 8. On the other hand, it can also be seen that the burst video traf®c corresponds to signi®cant delay jitter distribution that has longer tails across wide range (corresponding to large jitters) and lower peak values (corresponding to less probability for zero jitter). For example, as shown in Table 6, since the sports video stream with Std. of 2.2692 and Max./Mean of 6.3835 is more bursty than that of TalkÐShow video stream with Std. of 1.7158 and Max./Mean of 4.4420, it has 26.95% of frame delay jitter over more than 40 ms. By contrast, the TalkÐShow video stream only has 13.64% of frame delay jitter more than 40 ms. The MPEG includes a series of audio±visual standard known as MPEG-1 and MPEG-2 [5] which are the ®rst international standards in the ®eld of high-quality digital
audio compression. The following numerical results of inter-stream delay jitter between MPEG video and audio streams are evaluated using a CartoonÐAsterix as the tagged video stream and a 24 ms 1152 PCM audio clip with a sample rate of 48 kHz and encoded at 16 bits per sample. Fig. 12 presents the pdf of inter-streams delay jitter under the condition of different background traf®c burstiness. It can be seen that large background traf®c burstiness has more signi®cant effects on the distribution of inter-streams delay jitter with longer tails across wide range (corresponding to large jitters) and lower peak values (corresponding to less probability for zero jitter). For example, as shown in Table 7, when the background traf®c burstiness is 1.25, 21.49% of inter-streams delay jitters are ranging from 240 to 40 ms. On the other hand, when the background traf®c burstiness increases to 2.5, correspondingly, 51.05% of inter-streams delay jitters exceed 40 ms. Then the service quality is deteriorated seriously. 5. Conclusion Network delay and delay jitter are the important QoS parameters for real-time services, such as packet voice and packet video traf®c over the Internet. The delay jitter
Fig. 12. PDF of frame delay jitter between video and audio at different traf®c burstiness.
L. Zhang et al. / Computer Communications 25 (2002) 863±873 Table 7 Delay jitter distribution between video and audio at different burstiness Burstiness Jitter . 20 ms (%) Jitter . 40 ms (%) Jitter . 60 ms (%) 1.25 1.875 2.5
33.33 48.68 64.33
21.49 24.44 51.05
10.02 20.89 38.07
can be either positive or negative. A sequence of negative jitter (clustering) may result in downstream nodal congestion and consecutive packet loss. On the other hand, a sequence of positive jitters (dispersion) may result in consecutive packet signi®cant delays. The pdf of delay jitter provides high-order distribution of jitter length that demonstrates the jitter behavior in high-speed networks. Since the estimation of delay jitters is a dif®cult task due to its transient queuing effects, this paper focuses on the performance behavior of delay and delay jitter by taking into account of the dynamic nature of the background traf®c using transient queuing solution. The steady state solution of delay jitter is archived when the sequence of the tagged stream arrival goes to a large number. The obtained numerical results demonstrate the effects of delay and delay jitter on speech clips, video traces and interstreams between video and audio streams. It shows that network conditions including traf®c load, traf®c burstiness and burst-length all have signi®cant effects on the pdf of delay and delay jitters. It can be seen that the inter-streams delay jitter may in¯uence the service quality. On the other hand, large sequence of either clustered or dispersed packets have signi®cant effects on the performance of QoS. Since the effect of delay jitter on MPEG video stream can be removed using the playback buffer at the receiver end, the network engineers may need to pay more attentions to the effect of delay jitter on voice speech. References [1] MPEG traces available via anonymous FTP from ftp://ftp-info3. informatik.uni uerzburg.de/pub/MPEG/. [2] Telogy Networks, Voice over IP (VoIP), http://www.telogy.com. [3] Telogy Networks, Voice over Packet, White paper, http://www. telogy.com.
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