Comparative Study of Techniques to minimize packet loss in VoIP Shveni P Mehta

Transcription

1 Comparatve Study of Technques to mnmze packet loss n VoIP Shven P Mehta ABSTRACT Voce over IP s an upcomng technology that enables voce communcaton through the Internet. Packet-based network lnks are shared between dfferent connectons, whch gves rse to nteracton between varous traffc types. Excessve delay, packet loss, and hgh delay jtter all mpar the communcaton qualty. Although voce over IP s very economcal, people become hestant to use t due to the above-mentoned facts whch affect the voce qualty greatly. Delay jtter and packet loss are the two factors that affect the voce qualty the most. The delay jtter manfests tself as packet loss n the dejtter buffer, whch drops the late packets. It s mpossble to remove packet loss from the network but t can defntely be mnmzed. Several technques have been proposed to mnmze the effect of packet loss. Ths paper descrbes some of those technques and then compares the followng two technques on the bass of certan crtera: ) A New Technque for mprovng qualty of speech n voce over IP usng tme-scale modfcaton [] ) Adaptve playout schedulng and loss concealment for voce communcaton over IP Network [] Keywords VoIP, Packet loss, Adaptve playout, Tme-Scale Modfcaton.. INTRODUCTION For non-real-tme applcatons lke Telnet, FTP, and emal, TCP offers relable delvery of data, but for real-tme applcatons lke voce over IP, TCP s not approprate because t ntroduces too much delay n creatng that relablty. The three man problems occurrng n real-tme applcatons lke Voce over IP (VoIP) are: ) End-to-End delay: The total delay experenced by the packet from the sender tll t reaches the recever. ) Jtter: The varaton n packet nterarrval tme. The dfference between when the packet s expected and when t s actually receved s jtter. ) Packet loss: Loss of voce packets from sender to recever. The total packet loss s composed of two elements: ) packets lost over the network due to congeston, and ) packets arrvng late after ther expected playout tme that are dscarded by the recever. The jtter caused by varable delays n the network s ultmately translated nto the effect of packet loss n the network, as the packets arrvng after the playout tme are consdered as lost. Another factor that could affect the qualty of VoIP s the choce of codec used to transform and compress analog sgnals nto dgtal sgnals. Currently avalable codecs are lsted n Table. The Mean Opnon Score (MOS) s a measure to determne or compare the qualty of audo transmssons. It s wdely appled Table - Avalable Codecs [] Codec Standards Btrate G.7 µ-law G.7 A-Law G.76-7 ADPCM 5 bt G.76-7 ADPCM bt G.76-7 ADPCM bt G.78 GSM-FR G.79 G.7. 6 kbps 6 kbps 0 kbps kbps kbps 6 kbps kbps 8 kbps 5. kbps as a lstenng qualty scale. To evaluate a gven speech sample, humans lsten to a speech sample and evaluate t on a MOS Scale of to 5 as gven n Table. When comparng a speech sample aganst a reference speech sample, PESQ-MOS (Perceptual Evaluaton of Speech Qualty - MOS) or DMOS (Degradaton MOS) are determned on a scale of to 5 as gven n Table. Table - MOS, DMOS and PESQ-MOS Scale MOS Opnon DMOS or PESQ- MOS Opnon 5 Bad poor Far Good Excellent 5 Hgh bt rate Codecs Low bt rate Codecs Very annoyng annoyng Slghtly annoyng Audble but not annoyng Inaudble From [] t can be seen that the nfluence of samplng and dgtzaton s small but not neglgble. Lower bt-rate codecs show a larger degradaton of speech qualty. Codecs lke G.76-7 and G.78 are more susceptble to degradaton from lost packets than the other codecs lke G.7, GSM-FR, G.79 and G.7.. For hgh bt-rate codecs, packet loss values up to 0% are acceptable for good qualty, whereas for the low bt-rate codecs only -5% packet loss rate s acceptable because the number of samples lost wthn a packet covers more of the speech sgnal. However, packet sze tself does not have much nfluence on qualty for any of the codecs. For a connecton wthout a dejtter buffer, jtter has a sgnfcant nfluence on the perceved qualty of receved voce. For the connecton wth a dejtter buffer, both lost and delayed packets have the same effect n the dejtter buffer, whch drops the late packets. Thus, packet loss s the most mportant problem that should be studed to mprove the qualty of VoIP. st Computer Scence Semnar SB-T-

2 . BRIEF OVERVIEW OF SOME OF THE PROPOSED TECHNIQUES Many technques have been proposed to mnmze the effect of packet loss n VoIP. Ths secton descrbes some of the technques n bref to get an dea about varous approaches taken for mnmzng the effect of packet loss n VoIP.. Replacng Lost Packets Mayorga et al [] frst study the mpact of packet loss n dfferent transmssons wth respect to dfferent codecs and then propose reconstructon strateges to recover lost nformaton... Interleavng Ths technque dstrbutes the effect of the lost packets n order to reduce the mpact on qualty. The nformaton of a speech part s dstrbuted n multple packets. The data unts are regrouped n a crossed form before transmsson such that they are dstrbuted, and at the recever they are rearranged n ther orgnal form. Thus, nstead of losng the whole packet small parts from dstrbuted packets are lost, Fgure Lost Fgure. Example of Interleavng Before Transmsson Interleavng.. Repetton: Lost packets are replaced by copes of last receved packets... Smple Interpolaton: Conssts of nterpolatng (averagng) by usng the packets after and before the lost packet... Interleavng wth Repetton The data are nterleaved before sendng and then any mssng part s substtuted usng the repetton technque at the recever...5 Interleavng wth Interpolaton Calculaton The nterleavng technque s used before sendng and then the recever nterpolates to replace any mssng parts n the jtter buffer.. Forward Error Correcton and Concealment [7] Ths approach encompasses both loss correcton and loss concealment algorthms. Loss Correcton uses meda-dependent and meda-ndependent Forward Error Correcton (FEC) technques. FEC consttutes addng redundancy data to the normal voce stream to protect from packet loss. FEC ntroduces overhead n terms of the total amount of traffc on the network, but f the amount of redundancy s controlled then ths approach can be used. In meda-ndependent FEC, general protecton codes lke Reed Solomon or Vterb are used to produce an extra protecton packet that follows the protected set of voce packets. These codes do not depend on any partcular underlyng meda characterstcs, but ntroduce a hgher delay whch may not be tolerated by many applcatons, ncludng VoIP. In medadependent FEC, the sender uses a hgh-qualty codec to create the voce samples and a lower qualty codec to generate redundant bts that are added to every packet rather than beng sent n ther own separate packet. The recevng codec removes the redundancy. If the recever must use that redundant data to substtute for a lost packet, the result s a lower-qualty (but not mssng) segment of voce. It ntroduces mnmum delay but may ntroduce more computatonal processng delay as t s meda dependent. Concealment technques can be used to supplement FEC for even better lost-packet compensaton. The most common concealment approaches nclude: Slence substtuton s substtuton of the lost frame by a slence frame of the same temporal frequency, but t can ntroduce nose f several of them are ntroduced. In nose substtuton Gaussan nose frames are used to substtute for the mssng frames. Ths produces better qualty. In frame replcaton, mssng frames are replaced by already present redundancy n the voce. Ths has low computatonal complexty and s effcent as more redundancy s expected to be present n the neghborng voce frames. It does not need large temporal sze. Waveform substtuton uses the frames pror to the lost frames and tres to use the most recent ones. It examnes buffered frames and searches for the best match.. Optmzed unequal error protecton [5] Forward Error Correcton schemes allocate equal amounts of error-control resources to each voce packet, rrespectve of the perceved mportance of a packet. Ths technque proposes sgnaladaptve, unequal error protecton n whch certan packets are allocated more FEC protecton than others, dependng on ther perceved mportance. Here, the basc unequal protecton vares the number of copes of a packet that are pggybacked onto subsequent packets: an adaptve Reed Solomon (RS) codng scheme provdes only certan packets wth RS protecton. The error control resources that should be allocated to the packet are determned by antcpatng what the packet loss concealment wll do f the packet s lost, and calculatng the expected dstorton for dfferent protecton scenaros. Ths technque s based on Lagrangan optmzaton used n vdeo communcatons to balance rate aganst dstorton. st Computer Scence Semnar SB-T-

3 . Compressed Doman Packet Loss Concealment of Snusodally Coded Speech [6] For ths technque Rodbro et al [6] desgned a new codec where the speech sgnal s compressed at the transmtter usng a snusodal codng scheme workng at 8 kbps. At the recever the concealment s carred out workng drectly on the snusodal parameters, based on tme-scalng of the packets surroundng the mssng ones. Ths s essentally another type of nterpolaton approach..5 Tme Scale Modfcaton Approach [] In ths approach a tme-scale modfcaton algorthm has been used to mnmze the effect of packet loss wthout ntroducng addtonal delays. It provdes flexble arrval delay cut-offs to late arrvng packets by means of adaptve playout, whch helps n reducng the packet loss-rate at the recever..6 Adaptve Playout Schedulng and Loss Concealment [] In ths scheme the past statstcs on network delay are used to adaptvely adjust the playout tme of the voce packets. Contnuous speech samples are created at the recever usng tmescale modfcaton. Ths technque mproves the trade off between bufferng delay and late loss sgnfcantly.. EVALUATION OF TECHNIQUES The most mportant characterstcs n determnng a good soluton for VoIP nclude the followng: ) Recever-based technque, as they are faster and ndependent of the network delay characterstcs, and also have lower computatonal overhead at the sender or network []. ) Mnmzes overall delay and packet loss. ) Flexble arrval delay cut-offs for late arrvng packets, to reduce the packet loss-rate at the recever. ) Uses adaptve playout to mnmze overall delay and effects of lost packets. 5) Low complexty. 6) Mantans ptch frequences and ntellgblty of speech. 7) Technque tself ncludes the soluton for recoverng from burst losses. Of the technques descrbed above, the two that best meet these crtera and are smlar enough to be compared effectvely are: ) Tme-scale modfcaton Approach []. ) Adaptve Playout schedulng and Loss Concealment []. The specfc characterstcs to be consdered for comparson are: ) Recever-based ) Mnmzes overall delay and packet loss ) Flexble arrval delay cut-offs to late arrvng packets, reducng the packet loss-rate at the recever ) Consders both portons of packet lost,.e., packets lost n the network due to congeston at some ntermedate node and at the recever due to packets arrvng later than ther scheduled playout tmes 5) Generc and relatve computatonal overhead 6) Mantans ptch frequences and ntellgblty of speech 7) Soluton for burst losses The prmary reason for choosng the selected technques for comparson s that both are packet loss concealment technques based on tme scalng. Both also mantan an adaptve playout buffer at the recever to compensate for the jtter. A detaled comparson of these two follows.. Tme-scale Modfcaton Approach [].. GLS-TSM Algorthm Tme-scale modfcaton s the processng performed on speech sgnals that changes the perceved rate of speech wthout affectng the ptch or ntellgblty of the speech. GLS-TSM (Global Local Search Tme Scale Modfcaton) s a tme-scale modfcaton algorthm whch s an mproved verson wth low complexty of the Synchronzed Overlap and Add (SOLA) algorthm []. Let S s be the length of the output frame and S a be the length of the nput frame arrvng at the recever. The relaton between S s and S a s gven by S s = α * S a; where α s tme scale modfcaton factor (α < for compresson and α > for expanson). GLS-TSM algorthm was orgnally desgned for larger speech segments (S a of length 600 samples or 00ms) and a pror fxed value of α. VoIP uses smaller packet szes, hence the values of these parameters were redefned to match the requrement of shorter sgnal segments and per packet varablty of α []. GLS-TSM takes N samples from those arrvng at the recever and α s calculated from the arrval pattern at the recever on a packet-bypacket bass. GLS-TSM searches for the pont of best algnment n two steps. Frst t searches for global smlarty or smlarty over a tme nterval between the analyss and synthess frames by comparng the zero crossng rates. Then t searches for local smlarty about a sample pont. Once the pont of best algnment s located, the output sgnal y[n], s formed by fadng-n the analyss frame and fadng-out the synthess frame n the overlappng nterval, and then duplcatng the nput frame untl all N samples are exhausted []. 0-ms voce packets and playout buffer sze of 5 packets are used to create the llustraton n Fgure. Addtonal delay allowed per packet was set to ms and Tme- Scale Modfcaton (TSM) frame sze was set to 5 packets. In Fgure (a), the packet s delayed by 7 ms. Ths delay s accepted and the packet s played out by tme-expandng packet wth α =. and packets and wth α =.. So packet s avalable when playout of packet ends. But as 7 ms playout tme of packet s already taken away by packets,, and, packet s compressed wth α = 0.5 as n Fgure (b). In general f the stuaton arses where a packet s to be compressed wth α < 0.5, t s compressed down to α = 0.5 only and the rest of the compresson s shared by subsequent packets. Ths s to mantan synchronzaton wthout degradng voce qualty by excessvely compressng a packet. As a result, packet 5 s also compressed and played out wth α = 0.8. If packet does not arrve at ts rescheduled playout tme, t s supposed to be lost, Fgure (c). Hence, any packet wth sequence number > s searched. For buffer sze of fve packets, sequence numbers of 5 or 6 are searched. If packet 5 has arrved then t s expanded to compensate for resdual playout tme of packet. If expanson s st Computer Scence Semnar SB-T-

4 Seq. # (a) Seq. # Seq. # (b) Fgure. Workng of the proposed scheme wth packet delay jtter (a) and loss (b) more than.5 tmes, then subsequent packets share the compensaton to avod delverng poor qualty voce wth excessve expanson. In Fgure (c), packet 5 s expanded wth α =. to compensate for ms of packet. In the worst case, f both packets 5 and 6 don t arrve, then samples from the prevously successfully receved packet are repeated to compensate for resdual playout tme wth smoothng of the waveform occurrng at packet boundares. Ths reduces the number of fully-repeated packets and elmnates the waveform msmatch at packet boundares. Thus, acceptable delay for late arrvng packets and concealment of the lost packets s acheved effectvely by ths technque. The redefned parameter ranges of the GLS-TSM algorthm produced good qualty speech for 0.5 α.0. The nput speech materal was taken from the TIMIT database and fles were downsampled to 8 KHz and two fles were concatenated to create a reference speech fle of duraton 6-8 seconds []. The plots n Fgure show that for acceptable packet loss values between 5% and0% the voce qualty has been graded as.5 PESQ MOS and above, whch s between slghtly annoyng and audble but not annoyng. Flexble delay cut-offs and mproved loss concealment gve better qualty speech. The scheme s generc, computatonally effcent, and sutable for any practcal VoIP system.. Adaptve Playout schedulng and Loss Concealment for voce communcaton over IP Networks.. Adaptve Playout When a playout buffer s employed at the recever to absorb the delay jtter, there s a tradeoff between average tme spent n the buffer (bufferng delay) and the number of packets that have to be dropped because of late arrval (late loss). Increasng 5 6 PESQ Score PESQ Score PESQ Score RPL % RPL % p RPL % Fgure. MOS predctons for representatve nput fles wth random packet-loss percentages (no packet arrval jtter) [] bufferng delay wll mnmze packet loss but ncrease end-to-end delay, whereas decreasng bufferng delay wll decrease end-toend delay but ncrease packet loss. In ths technque adaptve playout s used to accommodate as many as possble late arrvng packets. In Fgure the graph shows the delay of voce packets on the network as dots and the total delay as a sold lne. Packets arrvng after the playout deadlne are lost and have to be concealed. As can be seen n the fgure, the playout s adjusted not only n slence perods but also wthn talk spurts accordng to the varyng delay statstcs, whch shows that the technque s hghly dynamc. Voce packets are scaled to mantan contnuous playout, whch ntroduces some playout jtter. However, ths flexblty allows reducng the average bufferng delay and late loss at the same tme. Hence, the tradeoff between bufferng delay and late loss s mproved. Ths technque s manly based on the average bufferng delay and the late loss rate as the basc performance measures. All measures are defned n Table. Assume that there are N packets n the packet stream. For voce, packets are fxed-sze blocks and outgong packets are generated perodcally. As shown n Fgure 5, the packets are sent from the st Computer Scence Semnar SB-T-

5 . Network Delay Total End-to-End Delay Fgure. Adaptve Playout schedulng scheme [] recever at tme t s at a constant packetzaton tme Lo; where =,,.N s the sequence number of the packet. These packets are receved at the recever at tme t r whch s gven by t + s - t s = Lo = constant. These packets are frst stored at the recever n a buffer for tme equal to bufferng delay and then played out at. tme t p Thus, db = t p - t r. The network delay of the packets can be gven by d n = t r - t s. Thus, total delay d t = d n + d b. If the packet s lost durng transmsson then d n =. Hence, the set of receved packets s gven by R = { t r < }. As can be seen from the Fgure 5, the adaptaton s performed on a packet-by-packet bass and each packet s played out wth dfferent lengths of the played out packets gven by L = t + p - t p ; where L s the acheved length (n tme) of audo packet. Average bufferng delay s gven by d b = (d max d n ); where P = { t p > t r } s the set of P played packets, and P denotes the cardnalty of the set. Late loss rate s gven by ε l = ( R P ). Lnk loss rate ε n = ( N R ). N N The total loss rate s the sum of late loss rate and lnk loss rate.e.; ε = ε l + ε n. Burst loss rate, denoted by ε b, s gven by ε b = ( B )/N; where B s the set of packets wth two consecutve losses and s gven by B = { t p < t r, t + p < t + r }... Scalng of voce packets wth WSOLA Waveform Smlarty Overlap-Add (WSOLA) decomposes the nput nto overlappng segments of equal length, whch are then realgned and supermposed to form the output wth equal and fxed overlap. The realgnment leads to a modfed output length. The segments to be added n overlap are searched on the bass of maxmum correlaton to ensure that they have the maxmum smlarty and the superposton wll not cause any dscontnuty before they are supermposed to generate smooth transtons n the output. Weghtng wndows are appled to the segments the reconstructed output. For speech processng, WSOLA has the advantages of mantanng the ptch perod, whch results n mproved qualty compared to resamplng. Sngle-Packet WSOLA The WSOLA approach has been modfed to work on only one packet so that ncomng packets can be scaled mmedately, wthout ntroducng any addtonal processng delay. To scale a voce packet, a template segment of constant length s selected from the nput. Next the segment of maxmum smlarty to the Notaton t s t r t p d n d b d b d t d max d max ε n ε l ε b ε R P B N L o L Sender Recever Playout Table Notaton [] Descrpton Tme packet s sent Tme packet s receved Tme packet s played out Network delay of packet Bufferng delay of packet Average bufferng delay of a stream Total delay of packet Fxed playout deadlne Playout deadlne of packet Lnk loss rate Late loss rate Burst loss rate Total loss rate Set of receved packets Set of played packets Set of packets lost consequtvely Number of packets n stream Sender packetzaton tme Actual length of scaled packet Packet Sequence Number t s d n L o t r d max t d p b L Fgure 5. Adaptve Playout [] Tme segment, called the smlar segment, s searched n the nput. The start of the smlar segment s searched n a search regon, as s shown n Fgure 6. For expandng short packets, the search regon s moved to the pror segment to fnd the frst smlar segment. Ths wdens the range of lookng for smlar waveforms. The pror packet could have been played out at the tme of scalng. Once the smlar segment s found, t s weghted by a rsng wndow and the template segment s weghted by a symmetrc fallng wndow. The smlar segment followed by the rest of the samples n the packet s shfted and supermposed wth the template segment to generate the output. Ths results n a long output as shown n Fgure 6(a). For example n Fgure 6(a), the output waveform has one extra ptch perod. The extra ptch perod s the nterpolaton of several ptch perods based on smlarty. If the output speech has not reached the desred length t s t r t p st Computer Scence Semnar SB-T-5

6 Ptch Perod 5 5 packet tme and results n better voce qualty. Waveform repetton s used to repar burst loss. Packet Packet Lost δl Smlar Segment Template Search Regon 5 Template Smlar Segment Found Smlar Segment Realgnment and Overlap Add Output / / / 5 Weghtng Wndow Search Regon 5 5 / / 5 Output + Extend to L o Extend to.l o (a) Lost Lost Output Extend to L o Extend to L o Extend to.l o (b) δl Lost Lost Output - + Extend to L o Waveform Repetton + Extend to.l o + + Tme δl (a) (b) Tme (Sample) Used for Correlaton Fgure 6. (a) Extenson and (b) compresson of sngle voce packets usng tme-scale modfcaton [] after such operatons, addtonal teratons are performed. Packet compresson s done smlarly, as shown n Fgure 6(b), but as we want the output of shorter length than the nput, the search regon for the smlar segment should not be defned n the pror packet. Packet compresson requres that a packet contans more than one ptch, whch lmts the mnmum length of the packet that can be compressed. However, a common packet length, such as 0 ms, s usually suffcent because ptch values below 00 Hz are not very frequent n speech sgnals. If for some reason packet compresson cannot be performed, then the compresson s shared by later packets. Ths scheme s entrely voce codec-ndependent. The begnnng and the end of each packet are not altered, so concatenaton of modfed packets needs no overlap to obtan smooth transtons. Hence, packets can be modfed ndependently and sent to the output queue back to back. To avod extreme scalng, L max =. L o and L mn = 0. L o, where L o s constant packetzaton tme.. Packet Loss Concealment In ths technque the packet loss concealment tres to cover late loss as well as lnk loss n the network by takng advantage of redundancy n the audo sgnal. It s a hybrd of tme-scale modfcaton and waveform substtuton. It scales one packet at a tme and uses two-sded nformaton along wth adaptve playout schedulng. Scalng one packet at a tme reduces the delay to one (c) Algnment determned by correlaton Fgure 7. Loss concealment for (a) sngle loss (b) nterleaved loss (c) consecutve loss [] As shown n Fgure 7, the packets are stored n a buffer at the recever after arrval. Packet s assumed lost f t s not receved by the tme packet - s to be played out, and the concealment starts at that moment. When packet assumed lost, ts pror packet - s extended wth a target length of L o and then played out. Further operaton depends on the loss pattern. If packet s the only packet lost and the followng packets are receved by ther deadlnes, packet + s extended wth a target length of.l o. As s shown n Fgure 7(a), small segments from packets - and + are searched for smlarty to obtan a mergng poston. The two segments are then weghted by fallng and rsng wndows before mergng. The total expanson of packets - and + can be bgger or smaller than the gap created by the lost packet. The resultng length of modfed packets s then L - + L + - L merge, and the playout tme of packet + s t p + = t p - + L - + L + - L merge For successful concealment t p + > t r + but n general, t p + wll not match the desred playout tme and t s lkely to be ether ahead or behnd the scheduled playout by a small dfference δl, as shown n Fg. 7. Ths dfference s corrected by the adaptve loss patterns or bursts loss can be covered as shown n Fgure 7(b) and 7(c), respectvely. In Fgure 7(b) when packet + s determned to be lost, packet + s scaled wth a target length of Lo nstead of. Lo to cover the gap resultng from the second loss n Fg. 7(c) when packets and + are lost, the waveform of the scaled packet - s repeated n order to conceal burst loss. In both cases, search of smlar waveforms s performed for mergng, and adaptve playout tme adjustment s used on the st Computer Scence Semnar SB-T-6

7 Trace STD of Network Delay (ms) Maxmum Jtter (ms) Table - Subjectve results of the technque [] Lnk Loss Rate (%) STD of Total Delay (ms) Bufferng Delay (ms) Total Loss Rate (%) Burst Loss Rate (%) MOS followng packet losses as shown n Fgure 7(c), waveform repetton s used for a maxmum of three tmes before the mechansm stops to generate output and resets the playout schedule. Burst loss degrades voce qualty most severely, even after beng concealed, because t s a smple repetton of pror waveforms.. Performance of the technque As gven n [], results were collected usng packet delay traces from the Internet by transmttng voce streams between hosts at four dfferent geographc locatons. The data sequences collected from these lnks were referred to as Traces respectvely. Subjectve testng for evaluaton of the qualty degradaton by scalng of voce packets on a scale of 5 to as gven n secton shows that the degradaton due to audo scalng s between naudble and not annoyng, even for extreme cases. In short, these results say that scaled audo has a good qualty. Subjectve testng to determne the overall qualty of speech usng adaptve playout n combnaton wth loss concealment was carred out usng four short segments from Traces that last for aproxmately 6s. The correspondng network characterstcs for these segments are gven n column of Table. No orgnal sample was provded for drect comparson, and the lsteners were asked to rate the qualty of speech usng an absolute scale of 5 to correspondng to 5-excellent, -good, -far, - poor, and -bad qualty, respectvely. The mean opnon scores (MOS) obtaned by averagng the scores from all lsteners and four dfferent samples are lsted n Table. In order to obtan reasonable sound qualty for each trace, the bufferng delay used s adjusted to the jtter characterstcs. Therefore, the hghest bufferng delay s used for Trace whle the bufferng delay for Trace s close to the mnmum of one packet tme (0 ms), whch s requred for loss concealment. Table data are qute detaled, coverng not just packet loss, but also network loss, jtter, bufferng delay, and burst loss. SUMMARY AND CONCLUSION The two technques under comparson may look very smlar on the whole, but when studed closely they have some dfferences between them. Table 5 captures the hghlghts of how Tme- Scale Modfcaton Approach [] and Adaptve playout schedulng and loss concealment [] compare. Thus, n ths paper two almost smlar technques have been compared. From Fgure and Table 5 t mght appear that Tme- Scale Modfcaton Approach [] s better from the pont of vew of MOS values, but the results presented for Adaptve playout schedulng and loss Concealment [] are more detaled. The latter technque uses traces from the real network as compared to the fles used n the former technque. Although Tme-Scale Modfcaton Approach [] s effectve, the Table 5. Comparson Table Tme-Scale Modfcaton Adaptve Playout Schedulng and loss concealment [] Approach [] Recever-based Mnmzes overall delay and packet loss Flexble arrval delay cut-offs to late arrvng packets, reducng the packet loss-rate at the recever, but t s lkely to ntroduce more delay as compared to the other technque because t s based on collectng N samples from the nput as descrbed n Secton.. but, ths technque s based on GLS-TSM algorthm as descrbed n secton.. where expanson and compresson of packets for tmescalng are based on α. The value of α cannot be more than.5 to mantan the qualty of sound. The workng of the algorthm shows that t s comparatvely less dynamc and s a smple tme-scale modfcaton technque., but t does not have to wat for N samples from the nput before processng starts and hence t has comparatvely less bufferng delay as demonstrated n. but, ths technque s based on WSOLA algorthm as descrbed n secton.. where expanson and compresson are based on fndng smlar waveforms rather than scalng the neghborng packets. The smlar segment s weghted by a rsng wndow and template segment s weghted by a fallng wndow. The two segments are then shfted and supermposed wth template to generate output. Ths shows that the technque s hghly dynamc. It detects arrval patterns from the nput and automatcally adjusts the amount of delay. The output s not smple tme-scale modfcaton but s based on smlarty and s a hybrd of tme-scale modfcaton and waveform substtuton. st Computer Scence Semnar SB-T-7

8 Consders both portons of packet loss,.e., packets lost n the network due to congeston at some ntermedate node and at the recever due to packets arrvng later than ther scheduled playout tmes 5 Generc and less computaton overhead 6 Mantans ptch frequences and ntellgblty 7 Soluton for burst losses Waveform Repetton. It ntegrates well wth the system, s ndependent of codec and the results demonstrate less computatonal overhead. Waveform Repetton Adaptve playout schedulng and loss Concealment [] gves a more detaled analyss based on total delay, bufferng delay and loss, burst losses, network delay and loss, hence t s more relable n terms of mnmzng packet loss along wth delay. The results presented by both technques are nsuffcent to compare them n terms of MOS and PESQ-MOS values. Adaptve playout schedulng and loss Concealment [] gves a detaled analyss for each trace but t s based only on a sngle loss rate value. If t had produced results for more MOS values usng vared packet losses, then the two technques could have been compared more effectvely. Ths leads to the need for further work n ths area. The other technques proposed so far for mnmzng packet loss can also be studed to determne whch technque best serves the purpose of mnmzng the effects n partcular network condtons. 5.0 REFERENCES [] Agnhotr, S., Aravndhan, K., Jamadagn, H.S., Pawate, B.I. A new technque for mprovng qualty of speech n Voce Over IP usng tme-scale modfcaton, Acoustcs, Speech, and Sgnal Processng, 00 Proceedngs. (ICASSP '0). IEEE Internatonal Conference on, Volume:, 00 Pages: [] Duysburgh, B., Vanhastel, S., De Vreese, B., Petrsor, C., Demeester, P. On the nfluence of best-effort network condtons on the perceved speech qualty of VoIP connectons, Computer Communcatons and Networks, 00 Proceedngs. Tenth Internatonal Conference on 5-7 Oct 00, Pages -9. [] Lang, Y.J., Farber, N., Grod, B. Adaptve Playout schedulng and Loss Concealment for voce communcaton over IP Networks, Multmeda, IEEE on Volume: 5, Issue:, Dec. 00 Pages: 5 5 [] Mayorga, P., Besacer, L., Lamy, R., Sergnat, J.-F. Audo packet loss over IP and speech recognton, Automatc Speech Recognton and Understandng, 00. ASRU '0. 00 IEEE Workshop on, 0 Nov.- Dec. 00, Pages: [5] Mngyu Chen, Murth, M.N. Optmzed unequal error protecton for voce over IP, Acoustcs, Speech, and Sgnal Processng, 00 Proceedngs. (ICASSP '0). IEEE Internatonal Conference on, Volume: 5, 7- May 00, Pages vol.5. [6] Rodbro, C.A., Chrstensen, M.G., Andersen, S.V., Jensen, S.H. Compressed doman packet loss concealment of snusodally coded speech ; Acoustcs, Speech, and Sgnal Processng, 00 Proceedngs. (ICASSP '0). 00 IEEE Internatonal Conference on, Volume:, 6-0 Aprl 00, Pages: I vol.. [7] Santos, P.M., Balbnot, R., Slvera, J.G., Castello, F.C. Analyss of packet loss correcton and concealment algorthms n robust voce over IP envronments, Communcatons, Computers and sgnal Processng, 00. ACRIM. 00 IEEE Pacfc Rm Conference on, Volume:, 8-0 Aug. 00, Pages:8-87 vol. -9 st Computer Scence Semnar SB-T-8