R1. What is meant by interactivity for streaming stored audio/video?What is meant by interactivityfor real-time interactive audio/video?

Medii de stocare streaming audio / video: pauz / resumat, re-pozitionare, repede-inainte, n timp real audio i video interactiv: oameni comunica si isi raspund in timp real. R2. Three camps were discussed for improving the Internet so that it better suports s multimedia applications. Briefly summarize the views of each camp. In which camp do you belong? Camp 1: Nu sunt schimbri fundamentale n protocoalele TCP / IP, se adaug limea de band n cazul n care este necesar, se utilizeaza, de asemenea, cache-ul, reelele de distribuie de coninut, i reelele de acoperire multicast. Camp 2: Furnizeaza un serviciu de reea care permite aplicaiilor s rezerve lime de band n reea. Camp 3, serviciu difereniat: introduce sisteme de clasificare i de securitate simple de la marginea reelei, i ofer diferite datagrame pe diferite niveluri de servicii n funcie de clasa lor n cozile router. R3. What are some typical compression ratios (ratio of the number of bits in an uncompressed object to the number of bits in the compressed versionof that object) for image and audio applications, and the compression techniques discussed in Section 7.1 ? Necomprimatele audio stocate pe CD au o rat de bii de 1411.2 Kbps. fiiere MP3 sunt de obicei codificate n 128 Kbps sau mai puin, conferindu-le astfel un raport de compresie de aproape 11. Ratele de compresie la imagini sunt n intervalul de 10 la 100. R4. Figures 7.1 and 7.2 present two schemes for streaming stored media. What are the advantages and disadvantages of each scheme? Figura 6.1: simplu, nu are nevoie de fiier meta sau server de streaming, Figura 6.2: permite playerului media sa interacioneze direct cu serverul web, nu are nevoie de un server de streaming; Figura 6.3: playerul media interacioneaz direct cu un server de streaming, care a fost conceput pentru aplicaii specifice de streaming. R5. What is the difference between end-to-end delay and packet jitter? What are the causes of packetjitter? ntrzierea end-to-end este timpul necesar unui pachet de a cltori n ntreaga reea de la surs la destinaie. ntrzierea de bruiaj este fluctuaia de end-to-end de ntrziere de la pachet pachetul urmtor. R6. Why is apacketthat is receivedafter its scheduledplayouttimeconsideredlost? Un pachet care ajunge dup jocul su programat n timp ce nu poate fi jucat. Prin urmare, din punctul de vedere al cererii, pachetul a fost pierdut. R7. Section 7.3 describes two FEC schemes. Briefly summarize them. Both schemes increase the transmissionrate of the stream by adding overhead. Does interleavingalso increase the transmission rate? Prima schem: trimite o bucat redundant codificata dup fiecare n buci; bucat redundanta este obinut exclusiv sau ING in n buci originale. Al doilea sistem: trimite un sistem de rat sczut de bii cu o rezoluie mai mic, mpreun cu fluxul original. Intercalare nu crete cerinele de lime ale banzii de flux.

R8. What is the role of the ONS in a CON? Does the DNShave to be modifiedto support a CON? What information, if any, must a CON provide to the ONS? Rolul DNS-ului este de a transmite cereri HTTP la serverul DNS care sunt gestionate de CDN, care, la rndul lor, redirecioneaz cererea pentru un server CDN adecvat. DNS-urile nu trebuie s fie modificate pentru a susine un CDN. Un CDN ar trebui s ofere DNS-ul cu numele de gazd i adresa IP a serverului de nume de autoritate (a se vedea seciunea 2.5.3). R9. What information is needed to dimension a network so that a given quality of service is achieved? a) Modele de cerere de trafic ntre punctele finale din reea b) cerinele de performan bine definite c) modele pentru care prezic performanta end-end pentru un model de volum dat, precum i tehnici de a gsi un cost ridicat de alocare minim de lime de band, care va duce la toate cerinele utilizatorilor ce sunt ndeplinite. R10. How are different RTP streams in different sessions identified by a receiver? How aredifferent streams from within the same session identified? How are RTPand RTPC packets (as part of the same session) distinguished? Fluxurile RTP n diferite sesiuni: sunt diferite adrese multicast, fluxurile RTP n aceeai sesiune: teren SSRC, pachetele RTP se deosebesc de pachete RTCP prin utilizarea numerelor de porturi distincte. R11. Three RTCPpacket types are described in Section 7.4. Briefly summarize the information contained in each of these packet types. Rapoartele de receptie a pachetelor: include informaii despre fraciunii de pierderi de pachete, ultimul numar de secventa, inter-sosire jitter; raportul de expediere a pachetelor: marca de timp i de blocul de timp de ceas din pachete RTP sunt cel mai recent generate, numrul de pachete trimise, numrul de bii trimii; descrie sursa pachete : adresa de e-mail a expeditorului, numele expeditorului, aplicaia care genereaz fluxul de RTP. R12. What is the role of a SIP registrar? How is the role of an SW registrar different from that of a home agent in Mobile IP? Rolul registratorului SIP este de a ine evidena utilizatorilor i a IP-urilor corespunztoare lor adrese care sunt utilizate n prezent. Fiecare registrator SIP ine evidena utilizatorilor care aparin domeniului su. De asemenea, nainte de a INVITA mesaje (pentru utilizatorii din domeniul su), la adresa IP pe care utilizatorul o folosete n mod curent. n acest sens, rolul su este similar cu cea a unui server de nume cu autoritate n DNS. R13. In Section 7.5, we discussed nonpreemptive priority queuing. What would be preemptive priority queuing? Does preemptive priority queuing make sens for computer networks? n ateptare prioritatii non-preemptive, transmiterea unui pachet nu este ntrerupt dup ce a nceput. n ateptare prioritatii preventive, transmiterea unui pachet va fi ntrerupt n cazul n care un pachet are prioritate mai mare ajunge nainte de finalizarea transmisiei. Acest lucru ar nsemna c poriuni ale pachetului va fi trimis n reea ca buci distincte; aceste buci nu ar mai avea toate cmpurile de antet corespunztoare. Din acest motiv, ateptare prioritatilor preventive nu sunt de folosit.

R14. Give an example of a scheduling discipline that is not work-conserving. O disciplin de programare care nu functioneaza conservant este o divizie de timp de multiplexare, prin care un cadru de rotaie este mprit n sloturi, cu fiecare fant disponibila exclusiv la o anumit clas. R15. Give an example from queues you experience in your everyday life of AFO, priority,RR, and WFQ. FIFO: linie la Starbucks. RR: fuzionarea de trafic (lund 1 vehicul din prima band si 1 din cea de a doua, apoi 1 de la primul, i aa mai departe). WFQ: Contor de bilete la aeroport pentru furnizarea de servicii pentru 2 persoane de la prima clasa si unul la clasa economica, i din nou 2 de la clasa nti, i aa mai departe. R16. What are some of the difficulties associated with the Intserv model and perflow reservation of resources? Scalabilitatea: rezervare de resurse Per-flow implic necesitatea unui router pentru a procesa rezerve de resurse i de a menine de stat per-flow pentru fiecare flux care trece prin router. Serviciu flexibil: Cadrul Intserv prevede un numr mic de clase de servicii prespecificate. P2. Consider the client buffer shown in Figure 7.3. Suppose that the streaming system uses the third option; that is, the server pushes the media into the . socket as quickly as possible. Suppose the available TCP bandwidth d most of the time. Also suppose that the client buffer can hold only about one-third of the media. Describe how x(1) and the contents of the client buffer will evolve over time. x (t), va impulsiona exponenial-rapid pana ce TCP-ul este disponibil pe limea de band, datorit TCP-ului nceput lent. Atunci x (t) va rmne aproximativ constant la TCP precum sidisponibilitatea de lrgimea de band pn la clientul tampon de umplere. La acea vreme, x (t), va scadea la aproximativ d pn cnd media de masa este trimisa i tamponul client rmne aproximativ complet. P3. Are the TCP receive buffer and the media player's client buffer the same thing? Ifnot, how do they interact? Nu, ei nu sunt acelai lucru. Aplicaia client citete date de la TCP unde primieste un tampon i pune-l n tamponul client. Dac buffer-client devine complet, atunci cererea se va opri la citirea din TCP primid un tampon pn la o anumit camer sa se deschida n tamponul client. P4. In the Internet phone example in Section 7.3. let h be the total numberof header bytes added to each chunk, including UDP and IP header. a. Assuming an IP datagram is emitted every 20 msecs, find the transmission rate in bits per second for the datagrams generatedby one side of this application. b. What is a typical value of h when RTPis used?

P5. Consider the procedure described in Sectiun 7.3 for estimating average delay dj Suppose that u = 0.1. Let rl -II be the most recent sample delay, let r2 -12 be the next most recent sample delay, and so on. a. For a givenaudio application suppose four packets have arrived at the receiver with sample delays "4- t4, r3 - '3' r2 - '2, and rl - 'I'Express the estimate of delay d interms of the four samples. b. Generalize your formula forn sample delays. c. For the formula in Part b, let n approach infinity and give the resulting formula. Comment on why this averaging procedure is called an.

C)

P6. RepeatParts a and b in Question 6 for the estimate of average delay deviation.

P7. For the Internet phone example in Section 7.3, we introduced an online procedure e (exponential moving average) for estimating delay. In this problem we will examine an alternative procedure. Let tj be the timestamp of the ith packet received; let rj be the time at which the ith packet is received. Let dn be our estimate of average delay after receiving the nth packet. After the first packet is received, we set the delay estimate equal to d, = r, -I,.

a. Suppose that we would like dn = (rl -11 + r2 - '2+ ... + rn - t,,)/n for all n. Give a recursive formula fordn in termsof dn_I, rn, and In' b. Describe why for Internet telephony, thedelay estimate described in Section 7.3 is more appropriate than thedelay estimate outlined in. Part a.

P8. Compare the procedure described in Section 7.3 for estimating average delay with the procedure inSection 3.5 for estimating round-trip time. What do the procedures have incommon? How are they different? Cele dou proceduri sunt foarte similare. Amndoua folosesc aceeai formul, ceea ce duce la scderea exponeniala a greuti pentru probele anterioare. O diferen este c pentru estimarea mediei RTT, n momentul n care datele sunt transmise i n cazul n care recunoaterea este primita este nregistrat pe aceeai main. Pentru estimarea ntrzierii, cele dou valori sunt nregistrate pe maini diferite. Astfel, ntrzierea de prob poate fi de fapt negativa. P9. Consider the adaptive playout strategy described in Section 7.3. a. How can two successive packets received at the destination have timestamps that differ by more than 20 msecs when the two packets belong to the same talk spurt? R. Daca este un pachet pierdut, alte pachete ar fi putut fi transmise in cele doua pachete. b. How can the receiver use sequence numbers to determine whether a packet is the first packet in a talk spurt? Be specific. R: S1 reprezinta numarul de pachete primite. pachetele I incep un nou transfer. Daca si atunci

P10. Consider the figure below (which is similar to Figure 7.5). A sender begins sending packetized audio periodically at t = I.The first packet arrives at the receiver at t = 8.

A. What are the delays (from sender to receiver, ignoring any playout delays) of packets 2 through 8? Note that each vertical and horizontal line segment in the figure has a length of I,2, or 3 time units. R. Intarzierea pachetului 2 este de 7 sloturi. Intarzierea pachetului 3 este de 9 sloturi. Intarzierea pachetului 4 este de 8 sloturi. Intarzierea pachetului 5 este de 7 sloturi. Intarzierea pachetului 6 este de 9 sloturi. Intarzierea pachetului 7 este de 8 sloturi. Intarzierea pachetului 8 este mai mare de 8 sloturi. B . If audio playout begins as soon as the first packet arrives at the receiver at t = 8, which of the first eight packets sent will not arrive in time for playout? R. Pachetele 3, 4, 6, 7 si 8 nu vor fi primite in timp util pentru transfer daca trabsferul incepe la t-8. C. If audio playout begins at t = 9, which of the first eight packets sent will not arrive in time for playout? R. Pachetele 3 si 6 nu vor fi primite la timp pentru transfer daca transferul incepe la t=9. D . What is the minimum playout delay at the receiver that results in all of the first eight packets arriving in time for their playout? R. Nici un pachet nu va fi transferat daca transferul incepe la t=10. P11. Consider again the figure in PIO,showing packet audio transmission and reception times. A . Compute the estimated delay for packets 2 through 8, using the formula for d, from Section 7.3.2. Use a value of u = 0.1 B. Compute the estimated deviation of the delay from the estimated average for packets 2 through 8, using the formula for v; from Section 7.3.2. Usea value of u = 0.1 T. Raspunsurile pentru punctele a si b sunt in tabelul de jos:

P12. Recall the two FEC schemes for Internet phone described in Section 7.3. Suppose the first scheme generates a redundant chunk for every four original chunks. Suppose the second scheme uses a low-bit rate encoding whose transmission rate is 2S percent of the transmission rate of the nominal stream. A . How much additional bandwidth does each scheme require? How much playback delay does each scheme add? R. Ambele scheme au nevoie de o marire de banda de 25%. Prima schema are o intarziere de 5 pachete. A doua schema are o intarziere de 2 pachete. B . How do the two schemes perform if the first packet is lost in every group of five packets? Which scheme will have better audio quality? R. Prima schema are capacitatea de a reconstrui codificarea initiala audio de inalta calitatea. A doua schema va folosi codarea audio de calitate scazuta pentru pachetele pierdute si, prin urmare, va avea calitatea de ansamblu mai mic. C . How do the two schemes perform if the first packet is lost in every group of two packets? Which scheme will have better audio quality?

R. Pentru prima schema, multe dintre pachetele originale se vor pierde i calitatea audio va fi foarte slaba. Pentru a doua schema, fiecare pachet audio vor fi disponibila pentru receptor, desi doar versiunea de calitate scazuta va fi disponibil pentru toate celelalte pachete. Calitatea audio va fi acceptabila. P13. Given that a CON does not increase the amount of link capacity in a network (assuming the CON uses existing links to distribute its content among CON nodes), how does a CON improve the performance seen by hosts? Give an example. R. Compania CDN furnizeaz un mecanism, astfel nct atunci cnd un client cere coninut, coninutul este furnizat de serverul CDN care poate furniza cel mai bine coninutul pentru un anumit client. Serverul poate fi cel mai apropiat server CDN la client (poate n acelai ISP ca i clientul) sau poate fi un server CDN cu o cale de congestie liber la client. n acest fel, chiar dac un CDN nu crete cantitatea de link n reea, performanele vzut de ctre gazde este mbuntit. P14. Is it possible for a CON to provide worse performance t.oa host requesting a multimedia object than if the host has requested the object from the distant origin server? Explain. R. Da. n cazul n care calea dintre gazda solicitant i server CDN, ales de CDN, este deja prea aglomerat sau dac serverul CDN ales este deja prea ocupat, atunci ar putea fi posibil ca gazda sa aiba experiene de performan joasa, ceea ce ar fi n cazul n care se contactateaza serverul de la distan n mod direct.

P15. How is the interarrival time jitter calculated in the RTCP reception report? (Hint: Read the RTP RFC.)

R. Bruiaj J de inter-sosire este definit ca fiind deviaia medie de diferen D n spaierea pachetului la receptor fa de expeditor pentru o pereche de pachete. Aa cum se arat n ecuaia de mai jos, acest lucru este echivalent cu timpul de tranzit relativ "pentru cele dou pachete; timpul de tranzit relativ este diferena dintre un pachet de RTP timestamp i ceasul receptorului n momentul sosirii. Dac este amprenta de timp RTP pentru pachete i i este timpul de sosire n RTP uniti timestamp pentru pachete i, apoi de dou pachete i i j, D este definit ca:

Bruiaj de inter-sosire este calculat n mod continuu ca fiecare pachet de date i care este primit, utiliznd aceast diferen D pentru acel pachet i pachetul anterior i-1 n ordinea sosirii (nu neaprat n ordine), conform formulei:

Ori de cte ori este emis un raport de primire, valoarea curent a J este eantionat.

P16. a. Suppose we send into the Internet two IPdatagrams, each carrying a different UOP segment. The first datagram has source IP address AI, destination IP address B, source port PI, and destination port T. The second datagram has source IP address A2, destination IP address B, source port P2, and destination port T. Suppose that A I is different from A2 and that PI is different from P2. Assuming that both datagrarns reach their final destination, will the two UOP datagrams be received by the same socket? Why or why not? R. Dup cum sa discutat n capitolul 2, sockets UDP sunt identificate de ctre cei doi - tuplu format din adresa IP destinaie i numrul de port de destinaie. Deci, cele dou pachete vor trece ntradevr, prin aceelai socket. b. Suppose Alice, Bob, and Claire want to have an audio conference call using SIP and RTP. For Alice to send and receive RTP packets to and from Bob and Claire, is only one UOP socket sufficient (in addition to the socket needed for the SIP messages)? If yes, then how does Alice's SIP client distinguish between the RTP packets received from Bob and Claire? R. Da, Alice are nevoie doar de un socket. Bob i Claire va alege diferit de SSRC, astfel Alice va putea face distincia ntre cele dou fluxuri. O alt ntrebare pe care am fi putut-o intreba este: Cum software-ul lui Alice tiu ce flux (de exemplu, SSRC) aparine lui Bob i care flux aparine lui Alice? ntr-adevr, software-ul lui Alice ar putea dori sa afieze numele expeditorului, atunci cnd expeditorul dicuta. Software-ul lui Alice devine SSRC de cartografiere pentru nume de RTCP sursa ce descriere rapoarte. P17. Consider an RTP session consisting of four users, all of which are sending and receiving RTP packets into the same multicast address. Each user sends video at 100 kbps. A . RTCP will limit its traffic to what rate? R. Limea de band pentru sesiune este de 4 * 100 kbps = 400 kbps. Cinci la suta din latimea de banda de sesiune este de 20 kbps. B . A particular receiver will be allocated how much RTCP bandwidth?

R. Fiecare utilizator trimis este atat un expeditor cat i receptor, fiecare utilizator primeste 5 kbps pentru pachete RTCP (rapoarte de recepie, rapoarte de expeditor, i descriere sursa de pachete). C . A particular sender will be allocated how much RTCP bandwidth? R. Fiecare utilizator trimis este atat un expeditor cat i receptor, fiecare utilizator primeste 5 kbps pentru pachete RTCP (rapoarte de recepie, rapoarte de expeditor, i descriere sursa de pachete). P18. A. How is RTSP similar to HTTP? Does RTSP have methods? Can HTTP be used to request a stream?

R. Cum ar fi HTTP, toate metodele de cerere i de rspuns sunt n text ASCII. RTSP are, de asemenea, metode (de configurare, redare, pauz), iar serverul rspunde cu coduri de rspuns standardizate. Da, utiliznd metoda GET, HTTP poate fi utilizat pentru a solicita un flux. B . How is RTSPdifferent from HTTP? For example, is HTTP in-band or out-of-band? Does RTSP maintain state information about the client (consider the pause/resume function)? R. Mesajele RTSP folosesc numere de port diferite de fluxuri media. Astfel, RTSP este out-of-band. Obiecte HTTP sunt trimise n mesajul de rspuns HTTP. Astfel, HTTP este in-band. HTTP nu ine de starea sesiune: fiecare cerere este tratat n mod independent. RTSP folosete ID-ul de sesiune pentru a menine starea sesiune. De exemplu, n laborator (sesiune de programare) pentru acest capitol, serverul RTSP este una de mai multe state pentru fiecare ID de sesiune. Atunci cnd n starea de pauz, serverul stocheaz numrul cadrului de la care a avut loc pauza.

P. 19 True or false: a. If stored video is streamed directly from aWeb server to a media player, then the application is using TCP as the underlying transport protocol. R. Adevarat b. When using RTP,it is possible for a sender to change encoding in the middle of a session. R. Adevarat c. All applications that use RTP must use port 87. R. Nu, streamurile RTP nu pot fi trimise/ primite de la orice numar de port. Vezi exemplu SIP in Sectiunea 6.4.3 d. If an RTPsession has a separate audio and video stream for each sender, then the audio and video streams use the Same SSRC. R. Nu. De obicei primesc alte valori SSRC diferite e. In differentiated services, while per-hop behavior defines differences in performance among classes, it does not mandate any particular mechanism for achieving these performances. R. Adevarat

f. Suppose Alice wants to establish an SIP session with Bob. In her INVITE message she includes the line: m=audio 48753 RTP/AVP3 (AVP3 denotes OSM audio).Alice has therefore indicated in this message that she wishes to send GSM audio. R. Fals. Indica ca doreste ca sa primeasca GSM audio. g. Referring to the preceding statement, Alice has indicated in her INVITE message that she will send audio to port 48753. R. Fals. Indica ca doreste sa primeasca audio pe portul 48753 h. SIP messages are typically sent between SIP entities using a default SIP port number. R. Adevarat. 5060 pentru ambele, sura si destinatii numerelor de port. I . . In order to maintain registration, SIP clients must periodically send REGISTER messages. R. Adevarat j. SIP mandates that all SIP clients support 0.711 audio encoding R. Fals, acesta este o cerinta de la H.323 si nu SIP.

P. 20 Suppose that theWFQ scheduling policy is applied to a buffer that supports three classes, and suppose the weights are 0.5, 0.25, and 0.25 for the three classes. A . Suppose that each class has a large number of packets in the buffer. In what sequence might the three classes be served in order to achieve the WFQ weights? (For round robin scheduling, a natural sequence is 123123123... ). R. O secventa posibila este 1 2 1 3 1 2 1 3 1 2 1 3 O alta secventa posibila este 1 1 2 1 1 3 1 1 2 1 13112113

B . Suppose that classes 1and 2 have a large number of packets in the buffer, and there are no class 3 packets in the buffer. In what sequence might the three classes be served in to achieve the WFQ weights? R. 1 1 3 1 1 3 1 1 3 1 1 3 P21. Consider the figure on the next page, which is similar to Figures 7.22-7.25. Answer the following questions

A . Assuming FIFO service, indicate the time at which packets 2 through 12 each leave the queue. For each packet, what is the delay between its arrival and the beginning of the slot in which it is transmitted? What is the average of this delay over all 12 packets? R:

B . Now assume a priority service, and assume that odd-numbered packets are high priority, and even-numbered packets are low priority. Indicate the time at which packets 2 through 12 each leave the queue. For each packet, what is the delay between its arrival and the beginning of the slot in which it is transmitted? What is the average of this delay over all 12 packets? R.

C . Now assume round robin service. Assume that packets I, 2, 3, 6, II, and 12 are from class I, and packets 4, 5,7,8,9, and 10 are from clatss 2. Indicate the time at which packets 2 through 12 each leave the queue. For each packet, what is the delay between its arrival and its departure? What is the average delay over all 12 packets? R:

D . Now assume weighted fair queueing (WFQ) service. Assume that oddnumbered packets are from class I, and even-numbered packets are from class 2. Class I hac;a WFQ weight of 2, while class 2 has a WFQ weight of l.Note that it may not be possible to achieve an idealized WFQ schedule as described in the text, so indicate why you have chosen the particular packet LO go into service at each time slot. For each packet what is the delay between its arrival and its departure? What is the average delay over all 12 packets? R.

e. What do you notice about the average delay in all four cases (FCFS, RR, priority. and WFQ)? R. Se poate observa ca intarzierea medie pentru toate cele patru cazuri este aceasi (1.91 seconds)

P.22. Consider again the figure for P21. A . Assume a priority service, with packets 1,4,5,6, and 11 being highpriority packets. The remaining packets are low priority. Indicate the slots in which packets 2 through 12 each leave the queue.

R:

B . Now suppose that round robin service is used, with packets 1, 4, 5, 6, and II belonging to one class of traffic, and the remaining packets belonging to the second class of traffic. Indicate the slots in which packets 2 through 12 each leave the queue. R.

C. Now suppose thatWFQ service is used, Withpackets 1,4,5,6, and 11 belonging to one class of traffic, and the remaining packets belonging to the second class of traffic. Class I has aWFQ weight of I, while class 2 has a WFQ weight of 2 (note that these weights are different than in the previous question). Indicate the slots in which packets 2 through 12each leave the queue. See also the caveat in the question above regardingWFQ service. R.

P23. Consider the figure below,which shows a leaky bucket policer being fed by a stream of packets. The token buffer can hold at most two tokens, and is initially full at I = O.New tokens arrive at a rate of one token per slot. The output link speed is such that if two packets obtain tokens at the beginning of a time slot, they can both go to the output link in the same slot. The timing details of the system are as follows:

1 . Packets (if any) arrive at the beginning of the slot. Thus in the figure, packets I, 2 and 3 arrive in slot O.If there are already packets in the queue, then the arriving packets join the end of the queue. Packets proceed towards the front of the queue in a FIFO manner. 2. After the arrivals have been added to the queue, if there are any queued packets, one or two of those packets (depending on the number of available tokens)will each remove a token from the token buffer and go to the output link during that slot. Thus, packets 1and 2 each remove a token from the buffer (since there are initially two tokens) and go to the output link during slot O. 3. A new token is added to the token buffer if it is not full, since the token generation rate is r = I token/slot. 4. Time then advances to the next time slot, and these steps repeat. Answer the following questions: A . For each time slot, identify the packets that are in the queue and the number of tokens in the bucket, immediately after the arrivals have been processed (step I above) but before any of the packets have passed through the queue and removed a token. Thus, for the t = 0 time slot in the example above, packets I, 2 and 3 are in the queue, and there are two tokens in the buffer. B. For each time slot indicate which packets appear on the output after the token(s) have been removed from the queue. Thus, for the t = 0 time slot in the example above, packets 1and 2 appear on the output link from the leaky buffer during slot O.

R.

P24. Repeat P23 but assume that r = 2. Assume again that the bucket is initially full. R.

P25 . Consider P24 and suppose now that r = 3, and that b = 2 as before.Will your answer to the question above change? R. Raspunsul este acelasi ca la problema 24

P26. Consider the leaky-bucket policer (discussed in Section 7.5) that polices the average rate and burst size of a packet flow.We now want to police the peak rate, p, as well. Show how the output of this leaky-bucket policer can be fed into a second leaky bucket policer so that the two leaky buckets in series police the average rate, peak rate, and burst size. Be sure to give the bucket size and token generation rate for the second policer. R.

P27 . Apacket flow is said to conformto a leaky-bucket specification (r,b) with burst size b and average rate r if the number of packets that arrive to the leaky bucket is less than rt + b packets in every interval of time of length t for all t. Will a packet flow that conforms to a leaky-bucket specification (r,b) ever have to wait at a leaky bucket policer with parameters rand b? Justify your answer. R. Nu

P28. Show that as long as rl < R w/(I.. w). then dlNJ. is indeed the maximumdelay that any packet in flow I will ever experience in theWFQ queue.

02. Do you think it is better to stream stored audio/video on top ofTCP or UDP? 03. Write a report on Cisco's SIP products. 04. Can the problemof providing QoS guarantees be solved simply by throwing enough bandwidth at the problem, that is, by upgrading all link capacities so that bandwidth limitations are no longer a concern? 05. An interesting emerging market is using Internet phone and a company's high-speed LAN to replace the same company's PBX (private branch exchange). Write a one-page report on this issue. Cover the following questions in your report: a. What is a traditional PBX? Who would use it? b. Consider a call between a user in the company and another user out of the company who is connected to the traditional telephone network.What sort of technology is needed at the interface between the LAN and the traditional telephone network? c. In addition to Internet phone software and the interface of Part b, what else is needed to replace the PBX? 06. Think about how the highway network is dimensioned (e.g., how the number of lanes in highways leading into and out of a big city is determined). List four steps that you think a transportation engineer takes when dimensioning such a highway.What are the analogous steps in dimensioning a computer network?

R . Sa fie timpul in care incepe sa se acumuleze trafic 1 in coada de asteptare. Ne referim la +t ca inceputul unei perioade ocupate de flux 1. Fie >t un alt timp in acelasi flux 1 a perioadei ocupate. Fie T1(,t) sa fie cantitatea de flux 1 de trafic transmis in intervalul ,* ,t]. Clar

Cum

Astfel, valoarea maxim a fluxului 1 de trafic n coada de ateptare

este. Rata minim de la care este servit acest trafic este

Astfel, ntrzierea maxim pentru un pic flux-1 este