This is a closed-book exam. You have 150 minutes to complete the exam. Please make sure that the problems appear in order in your "blue book", with each problem starting on a new page. The exam has a total of 62 points.
See class notes and RFC 2205, section 2.5: "The first killer reservation problem (KR-I) arises when there is already a reservation Q0 in place. If another receiver now makes a larger reservation Q1 > Q0, the result of merging Q0 and Q1 may be rejected by admission control in some upstream node. This must not deny service to Q0." [3 points for this part]
The problem is caused by the merging of different reservations coming from receivers for the same flow and appears in any receiver-oriented reservation protocol with diverse receiver requirements. It is not caused by having small and large reservations for different flows - large flows blocking smaller ones is a generic problem of any reservation protocol. 
[need 3 of the above]
property E.164 IP DNS format textual, numbers, < 15 digits 32-bit binary dot-separated text assignment geographic, function (800) network topology/provider, function (multicast) organization structure fixed (NANP: country, area code, exchange) variable-length prefix country/type, organization, sub-organization capacity > 1015 109 > 1028 (with 16 letters) mapping 800 to landline, directory service ARP: IP to MAC, RARP: MAC to IP DNS: DNS to/from IP density very high, with area-code splits high, but permament very low sharing of names several home phones = one number; hunt groups each interface = one IP several hosts = one name; one host = several names local names yes, several levels no, but net 10 + NAT yes, within domain
[Need at least 2 properties.]
property ADSL cable modem downstream bandwidth lower (1.5 Mb/s) higher (30 Mb/s) QoS dedicated pipe shared bandwidth upstream bandwidth lower (16...640 kb/s), but fixed; sufficient for video? lower than downstream (768 to 2 Mb/s, shared) multicast possible, but no efficiency gain very efficient; downstream only Internet telephony dedicated access bandwidth, may not be sufficient for video shared, low upstream bandwidth (typically, engineered for 150 kb/s max.); downstream engineered to guarantee at least ISDN (128 kb/s) speed
Header length, header checksum, fragmentation.
The congestion window increases by one for each packet received (i.e., doubles every RTT) until the packet loss occurs. Note that the packet loss occurs after 12 packets, not 12 round-trip times. After the 12th packet is lost, the sender will get duplicate ACKs and reset ssthresh and cwnd to half the current congestion window.
All group members are scheduled to transmit their first RTCP in the 2.5 second interval between 1.25 and 3.75 seconds. 0.1/2.5 of the users, or 400, will send a packet before they have received a packet from anybody else. For this initial burst, reconsideration cannot help.
Note: Due to imprecision in question, have to accept "10,000" as an answer since the total number of packets transmitted will be that high.
One-way delay coast-to-coast is approximately 30 ms. We assume that no packets are lost. For a redirect server, there would be two trips across the country, for a total delay of 150 ms. The second INVITE would require the caller to wait for the ACK, thus the extra 30 ms.
For a proxy server, we can ignore the propagation delay within the CUCS domain. The call would take about 90 ms to set up.
Processing overhead is somewhat less for a redirect server since it does not have to generate its own call.
Packetization overhead; double talk; "hang-over" at the end of words and sentences; background noise. [Any two]
[2 pts. each; if factors such as bandwidth share, hop count, etc. are listed, 1 pt. each]
- Buffered packets due to burstiness; depends on bucket depth and rate share.
- Transmission delay caused by bandwidth share, depends on hop count, bandwidth share and maximum own packet length.
- Interference by other packets, depends on number of hops and max. length of all packets.
A structural/presentation oriented version would use HTML, e.g.,<h2>John Doe</h2> <dl> <dt>School: <dd>FUSEAS <dt>Year of graduation: <dd>2002 <dt>GPA: <dd>2.9 </dl>
For XML: see class notes.<student> <school>FUSEAS </school> <graduation>2002 </graduation> </student>
CSS: see class notes. [2 pts. each for HTML, XML and CSS.]
This can be accomplished within HTTP using server-driven Content-negotiation using the HTTP Accept-Language header or 300 Multiple Choices. Caches need to keep track of the Content-Language or ETag field to make sure they return the correct version. Apache, for example, supports having a single URL return two different files.
As an alternative, cgi-bin scripts could be used to either return the appropriate page or redirect the client to the right URL. The script would read the Accept-Language header and return the appropriate version. However, for appropriate caching, it must set the same Content-Language or ETag fields as in the server-supported solution. Redirects are less appropriate due to the additional round-trip delays.
A page with clickable links is not as convenient.
URNs (together with URCs) could be used, but not in this question.
Assume first, that this radio program is distributed via unicast, as in today's RealAudio.
With SIP, the radio receiver would "invite" (using INVITE) the radio station when joining a new subnet (with a new unicast address) and send a BYE request when leaving the subnet. Alternatively, the sender could also use the receiver's RTP reports to discover the new location. Having the radio server issue an invitation is inappropriate since the server doesn't know the identity of radio receivers.
SIP does not have a "redirect" request (RTSP does). Registration is not helpful, as it only affects new, server-initiated invitations.
Another design alternative is to announce the radio program using SAP/SDP, and then receive the program using multicast. This requires that join latencies are low enough. Any delay in canceling the transmission will lead to unnecessary traffic in the subnet that the receiver just left.
Another design alternative would be RTSP, using the REDIRECT request.