Fakultt Informatik Institut fr Systemarchitektur, Lehrstuhl Rechnernetze

Internet Services & Protocols

Solution 4: Multimedia Applications
1. What is the difference between end-to-end delay and packet jitter? What are the causes of packet jitter? End-to-end delay is the time it takes a packet to travel across the network from source to destination. Delay jitter is the fluctuation of end-to-end delay from packet to the next packet. 2. Jitter Compensation: Why is a packet that is received after its scheduled playout time considered lost? A packet that arrives after its scheduled playout time can not be be played out. Therefore, from the perspective of the application, the packet has been lost. 3. Briefly summarize MQS and Interleaving. How do both differ concerning the transmission rate? MQS (Mixed Quality Streams): send a lower-resolution low-bit rate scheme along with the original stream. Interleaving: break down a specific packet into several smaller parts and interleave them before sending. Interleaving does not increase the bandwidth requirements of a stream as it does not introduce any redundancy. 4. How are different RTP streams in different sessions identified by a receiver? How are different streams from within the same session identified? How are RTP and RPTC packets (as part of the same session) distinguished? RTP streams in different sessions: different multicast addresses; RTP streams in the same session: SSRC field; RTP packets are distinguished from RTCP packets by using distinct port numbers. 5. True or false: 5.1 When using RTP, it is possible for a sender to change encoding in the middle of a session. True. 5.2 All applications that use RTP must use port 87. No, RTP streams can be sent to/from any port number. 5.3 Suppose an RTP session has a separate audio and video stream for each sender. Then the audio and video streams use the same SSRC.

No, typically they are assigned different SSRC values. 5.4 Suppose Alice wants to establish an SIP session with Bob. In her INVITE message she includes the line: m=audio 48753 RTP/AVP 3 (AVP 3 denotes GSM audio). Alice has therefore indicated in this message that she wishes to send GSM audio. False, she is indicating that she wishes to receive GSM audio. 5.5 Referring to the preceding statement, Alice has indicated in her INVITE message that she will send audio to port 48753. False, she is indicating that she wishes to receive audio on port 48753. 5.6 SIP messages are typically sent between SIP entities using a default SIP port number. True, 5060 for both source and destination port numbers. 5.7 In order to maintain registration, SIP clients must periodically send REGISTER messages. True.