Assignment
TCOM 590, Fall 2013
Homework #1
Due on September 24th
Delivery method: hard or soft copy (email is just fine)
Dharanidhar
G00817520
1. Assume audio conferencing session with multiple participants (RTP streams). How would you:
a. Identify deferent voice streams that belong to the same session?
b. Differentiate between RTP and RTCP packets that belong to the same session?
c. Conclude if there is as media mixer in this session?
d. How many voice milliseconds (ms) have been packetized if the RTP timestamp for the first packet, R1 = 980 and the RTP timestamp for the n-th packet Rn =
2100?
e. Indicate the beginning of talkspurt?
f. Reorder incoming RTP packets at the receiver end?
g. Determine the voice payload size if PCM is used to digitize voice signal?
h. Detect RTP packet loss?
(16 points)
a. The SSRC in the RTP header helps identify voice streams that belongs to the same source in the session. The RTP SSRC helps label streams from different sources.
b. If the port number is even then the packet is an RTP packet. If the port number is odd then the packet is an RTCP packet. Usually RTP and
RTCP(RTP+1) are assigned UDP ports from 16382-32500
c. Yes, there should and must be a media mixer in the conference session. The mixer essentially receives RTP streams from one or more sources, combines them in some way and forwards the new mixed stream to one or more receivers. d. Timestamp to timevalue conversion : Rn-Ri /8000Hz
=) (2100-980)/8000 = 140µsec
Therefore, packets of size 140 µsec have been packetized.
e.
The beginning of talkspurt is indicated by the marker bit. It also helps adjust the playout delay at the receiver and is also used for silence suppressions to indicate packets that follow periods of silence.
f. Sequence number in the RTP header helps reorder the RTP packets at the receiver end. And the timestamp helps play out the voice packets in sequence. g. The payload type field in the RTP header indicates the
Homework #1
Due on September 24th
Delivery method: hard or soft copy (email is just fine)
Dharanidhar
G00817520
1. Assume audio conferencing session with multiple participants (RTP streams). How would you:
a. Identify deferent voice streams that belong to the same session?
b. Differentiate between RTP and RTCP packets that belong to the same session?
c. Conclude if there is as media mixer in this session?
d. How many voice milliseconds (ms) have been packetized if the RTP timestamp for the first packet, R1 = 980 and the RTP timestamp for the n-th packet Rn =
2100?
e. Indicate the beginning of talkspurt?
f. Reorder incoming RTP packets at the receiver end?
g. Determine the voice payload size if PCM is used to digitize voice signal?
h. Detect RTP packet loss?
(16 points)
a. The SSRC in the RTP header helps identify voice streams that belongs to the same source in the session. The RTP SSRC helps label streams from different sources.
b. If the port number is even then the packet is an RTP packet. If the port number is odd then the packet is an RTCP packet. Usually RTP and
RTCP(RTP+1) are assigned UDP ports from 16382-32500
c. Yes, there should and must be a media mixer in the conference session. The mixer essentially receives RTP streams from one or more sources, combines them in some way and forwards the new mixed stream to one or more receivers. d. Timestamp to timevalue conversion : Rn-Ri /8000Hz
=) (2100-980)/8000 = 140µsec
Therefore, packets of size 140 µsec have been packetized.
e.
The beginning of talkspurt is indicated by the marker bit. It also helps adjust the playout delay at the receiver and is also used for silence suppressions to indicate packets that follow periods of silence.
f. Sequence number in the RTP header helps reorder the RTP packets at the receiver end. And the timestamp helps play out the voice packets in sequence. g. The payload type field in the RTP header indicates the