1.
Assuming standard 1500 byte Ethernet max payloads: how many IPv4 fragments will be needed to
transfer 2000 bytes of user data with a single UDP send? And, how do the 2000 bytes get split over the
frags?
Answer:
We first need to factor in the space needed for the IP header, which is20 bytes, and the UDP
header, which is 8 bytes. This is a total of 28 bytes that need to be factored in immediately, which
leaves a total of 1472 bytes for data (PCVR.NL, N.D.). This allows us to figure the number of
fragments that will be needed for 2000 bytes of date, which is 2. The first fragment will contain a total
of 1500 bytes of data, which will include the 20 bytes for the IP header, the 8 bytes for the UDP
header, and 1472 bytes of user data. The remaining 528 bytes of user data will be the total amount
of data in the second fragment, as it will not include either the IP header or UDP header
2. Despite its conceptual elegance, RPC (Remote Procedure Call) has a few problems. Discuss any 3 of
those in brief.
Answer:
->The first issue is Marshalling: RPC frameworks have a stub for the functions involved. Before
sending the message the stub has to Marshall the parameters so they can be sent across the
network. Parameters consist of simple types and compound types. Due to each type having its own
representation, the types of the various parameters must be known to the modules that actually do
the conversion. Without that knowledge, conversations can’t start.
? The second issue is Lack of Streaming: This problem deals with the fact that when the server is
building the response it must build it entirely and the client is then idle for that period.
?The third issue is Single Threaded Servers: RCP favors the use of multi-threaded servers instead
of single threaded ones. This is due to the RPC model not allowing the server to return without
serving a response to the client.
3. Why is timestamping needed in real-time applications? This is in the context of Real-time Transport
Protocol (RTP).
Real-time Transport Protocol RTP:: is used for transmission of real time data like audio and videos.
Jitter: It means variable delays in packet transmission. Mostly seen in audio and video data
transmissions. But we want to reduce this jitter effect.
So, RTP will compensate that jitter effect so that user won't anguish. For that Time stamps and
sequences numbers are being used here. Sequence numbers are used for reliability of packet
transmission whereas timestamp is used to place the incoming audio and video packets in the
correct timing order (playout delay compensation)....
4. Why does UDP exist? Would it not have been enough to just let user processes send raw IP packets?
Answer:
By using UDP, a segment will be delivered correctly to the specified application because UDP uses
source and destination ports while raw IP packet does not include ports. That is, a segment cannot
be delivered to a specified application as a raw IP packet.
5. Explain how QUIC eliminates a couple of RTTs usually needed at the start of a secure
web connection.
QUIC work as an application protocol over User Datagram Protocol (UDP). It integrates
ideas from Transmission Control Protocol (TCP), TLS, and Datagram Transport Layer
Security (DTLS) to provide security comparable to TLS and minimal latency during
connection setup.
QUIC provides security by encrypting the application data and most of the protocol header. It
provides improved latency with 0–RTT connection establishment. A client that has previously
communicated with a server can start a new connection without a three-way handshake and
enables to send useful data in the very first round trip. The reduction in initial latency in
comparison to TLS equals two or three round-trip times (RTTs), depending on TLS session
resumption is enabled. QUIC improves upon TCP and TLS in other ways also such as
including introduction of multiple streams per connection to reduce head-of-line blocking and
vastly improved acknowledgment information that eliminates retransmission ambiguity
issues.
