Q1 To maintain a seamless VoIP conversation, the end-to-end delay must be kept below:
A 50 ms B 150 ms C 400 ms D 1000 ms
Q2 The fluctuation in end-to-end delay of consecutive packets caused by variable network congestion is known as:
A Packet loss B Propagation delay C Delay jitter D Interleaving
Q3 How does increasing the playout buffer (q q q ) affect VoIP performance?
A Improves interactivity and reduces packet loss. B Worsens interactivity but reduces packet loss. C Improves interactivity but increases packet loss. D Worsens interactivity and increases packet loss.
Q4 Interleaving is a loss concealment technique that works by:
A Sending duplicate packets. B Requesting retransmission of lost packets via TCP. C Spreading chunks of audio across multiple packets to prevent complete gaps. D Reducing the audio encoding rate dynamically.
Q5 Which of the following is true regarding the Real-Time Transport Protocol (RTP)?
A It provides guaranteed Quality of Service (QoS). B It is typically encapsulated in TCP segments. C It provides a standardized packet format including sequence numbers and timestamps. D It acts as an out-of-band control protocol for VCR functions.
Q6 Which RTP header field allows the receiver to properly synchronize playout and handle jitter?
A Payload Type B Sequence Number C Time Stamp D SSRC
Q7 What is the primary function of the Real-Time Control Protocol (RTCP)?
A To carry the audio/video payload. B To limit total bandwidth usage to 50%. C To provide periodic transmission statistics and synchronize streams. D To map SIP URIs to IP addresses.
Q8 RTSP (Real-Time Streaming Protocol) is primarily designed to:
A Provide "Internet VCR" remote controls for streaming media servers. B Setup VoIP phone calls. C Bypass NAT restrictions. D Distribute files in a P2P network.
Q9 In SIP, which server is responsible for routing INVITE messages to locate the callee?
A Tracker B Registrar Server C Proxy Server D Supernode
Q10 What message is used by a SIP client upon bootup to bind its current dynamic IP address to its SIP URI?
A SETUP B INVITE C HELLO D REGISTER
Q11 If a SIP callee receives an INVITE but does not support the requested media codec, what response should they send?
A 200 OK B 606 Not Acceptable C 404 Not Found D ACK
Q12 In a P2P network distributing a file to N N N peers, what happens to the total available upload capacity as N N N increases?
A It decreases. B It remains constant. C It increases because new peers bring additional upload capacity. D It fluctuates randomly.
Q13 What is the role of a Tracker in the BitTorrent protocol?
A It stores the rarest chunks of the file. B It tracks the IP addresses of peers participating in a torrent. C It enforces the Tit-for-Tat algorithm globally. D It converts the file into 256 Kb chunks.
Q14 BitTorrent's "Rarest First" strategy is designed to:
A Download the smallest chunks first. B Ensure that the least common chunks are duplicated quickly to prevent data loss if the original seeder leaves. C Send data only to peers with high download speeds. D Encrypt the chunks with the rarest keys.
Q15 The BitTorrent algorithm that rewards peers for uploading and penalizes leechers is called:
A Rarest First B Optimistic Unchoking C Tit-for-Tat D Supernode Routing
Q16 What is the purpose of Optimistic Unchoking in BitTorrent?
A To randomly disconnect from slow peers. B To probe for better trading partners and give brand-new peers a chance to get their first chunk. C To unchoke the Tracker server. D To bypass NAT routers.
Q17 Classic Skype utilized a hybrid architecture consisting of:
A P2P authentication and centralized media routing. B Centralized login/authentication and P2P user discovery/routing. C Centralized SIP Proxies and centralized Media Servers. D Pure P2P with no centralized servers at all.
Q18 What is a Skype Supernode?
A A Microsoft datacenter server used for billing. B A standard peer hidden behind a strict NAT router. C A highly capable peer with a public IP that acts as a routing hub in the overlay network. D A specialized BitTorrent tracker.
Q19 How does Skype overcome the issue of NAT Traversal when both callers are behind NAT routers?
A It uses a centralized SIP proxy to route the call. B It forces the routers to open port 5060. C Both peers maintain open outbound connections to Supernodes, which then act as traffic relays. D It disables the firewalls using an RTSP SETUP command.
Q20 During a talk spurt, if audio is encoded at 64 kbps and packetized every 20 msec, what is the size of the payload chunk in bytes?
A 128 bytes B 160 bytes C 256 bytes D 320 bytes