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Multimedia
Communications over the
Internet
IP Packet-Switching Networks
• Packet-switching protocols based on the Internet
Protocol (IP) generally consist of a variety of
different subnetworks of various technologies
– The datalink and physical layer are largely outside
the scope of the Internet efforts
• IP is a connectionless (datagram) network layer
– Different packets may traverse various routes
between source and destination (they may arrive
out of order or be duplicated)
– Packets are transmitted on a best effort basis (no
guarantee that a packet will arrive at destination
IP Networks (cont’d)
• IP provides for the transmission of blocks of data
called datagrams from source to destination,
where source and destination are hosts identified
by fixed-length addresses
• Datagrams can be as large as 64 KB, but usually
the are ~1500 bytes long.
• IP also provides for fragmentation and reassembly
of long datagrams, if necessary, for
transmission through small packet networks.
• IP basically runs over any Media Access Control
(MAC) protocol
Internet Protocol Relationships
(Non Real-Time)
• At the highest level, the users invoke application programs
that access services available across the Internet. An
application interacts with the transport-level protocol to
send or receive data.
• Each application program chooses the size of transport
needed, which can be either a sequence of individual
messages or a continuous stream of bytes.
IP Relationships (cont’d)
• Above IP, there are two transport layer options:
TCP and UDP.
– TCP provides end-to-end communication. It takes
care of reliable, error-free transfer of data, and insequence delivery. UDP has less overhead
compared to TCP, but does not guarantee
transfers.
– Both protocols support multiplexing, i.e. they allow
several distinct streams of data between two hosts
• Streams are labeled by source and destination port
numbers
UDP
• The User Datagram Protocol (UDP) offers only
minimal services beyond those provided by the
network layer
– Multiplexing via port number
– Checksum for the payload in the packet
• IP provides only a checksum for the IP header
TCP
• The Transmission Control Protocol (TCP) offers a
reliable, sequenced byte stream service between
two Internet hosts
• Reliability is achieved by retransmission of lost or
erroneous packets
– Requires acknowledgment of received data
• Flow control is achieved by means of a sliding
window mechanism
– Also used for congestion control: the sender reduces
the window size in the case of traffic congestion
• The sender probes for the currently available bandwidth by
gradually increasing its window size until it senses packet loss - at
which point it quickly reduces the window size
TCP (cont’d)
• TCP is less suited to multimedia data than UDP
– It trades reliability for delay jitter
– Since it enforces in-order delivery, a single lost packet
can hold up packets arrived after it
– To improve delay behavior, an implementation is
free to accept out-of-order packets and to
acknowledge packets that have not been received
but are excessively delayed
• However, this may interfere with the TCP congestion control
– The TCP congestion control mechanism imposes
throughput limitations that may change on very short
time scales
Connectivity Requirements
• Connectivity requirements examples
– Accommodating a listening audience of 25,000
streams daily with an average listening time of 6
minutes per stream requires, on average, only
100 concurrent streams
– Accommodating a daily listening audience of
250,000 streams, with an average listening time
of 20 minutes per stream, requires an average of
6,000 concurrent streams
Connectivity Requirements
(cont’d)
• Due to variable traffic patterns, load is not
constant throughout the day, therefore the peak
number of streams required may be twice or three
times the average
• When implemented, broadcast protocols (to an
entire subnetwork) or multicast protocols (to
members of a multicast group in various scattered
subnetworks) allow to reduce the load on the
network by allowing many user to share a single
stream from the server
IP Multicast
• IP Multicast allows a sender to transmit an IP
packet to multiple receivers. Three possible ways for
multicast:
– Setting up multiple virtual circuits (not possible in Internet)
– Including list of addresses in packet header (what happens if
1,000,000 addresses?)
– Radio-like approach (used by IP Multicast)
• Sender chooses an IP address from the class-D set (244.0.0.0 to
239.255.255.255). Receivers subscribe to multicast finding out the
multicast address.
• An IP multicast group can have any number of senders and
receivers. Membership is dynamic.
• IP Multicast is independent of transport mechanism
(but TCP cannot be used)
HTTP
‧ HTTP (Hyper-Text Transfer Protocol) is built on
top of TCP/IP
‧ For transmission through firewalls, often
multimedia must be encapsulated in a HTTP
envelope, lacking most of the required real-time
features
‧ Multimedia data is downloaded on the client
computer. Fast-start techniques can be used to
begin playback as soon as enough of the
content has been downloaded to the client
• UDP and TCP protocols are otherwise used
– UDP is sometimes blocked by firewalls
Requirements for Real-Time Multimedia
• Sequencing. Packets must be reordered in real
time at the receiver. If a packet is lost, must be
detected and compensated without retransmission
• Intramedia synchronization. Need some form of
“time stamping” to know when to playback packets
– Very important for VBR traffic
• Intermedia synchronization. E.g., audio must be
synchronized with video (lip-sync).
• Payload identification. E.g., for media filtering
• Frame indication. For synchronized delivery, it is
useful to indicate when a video frame (or audio
segment) begins or end.
RTP
‧ RTP (Real-Time Protocol) is a transport
protocol for audio and videoconferences and
other multiparticipant real-time applications.
‧ Designed to run over multicast IP
‧ Light-weight protocol, without error
correction, flow control, or guaranteed time
delivery functionality.
‧ Offers services such as playout synchronization,
demultiplexing, media identification, and activeparty identification.
Functionalities of RTP
• Re-sequencing and loss detection
• Multicast-friendly
• Media-independent (voice, video,…)
• Explicit support for mixers and translator
– Mixers: take media from several users and
mix them into one media stream
(e.g., conference bridge)
– Translators: take a single media stream,
convert it to another format, and send it out
(e.g. media filtering)
Functionalities of RTP (cont’d)
• QoS feedback (via RTCP)
– RTP sources can use this information to adjust their
data rate (media scaling)
• Loose session control
– Using RTCP, participants can periodically distribute
identification information (name, e-mail address,…)
– Provides awareness of who is participating in a
session without maintaining a centralized
conference participant registry
• Encryption
RTP (cont’d)
• RTP specifies a packet structure for packets
carrying audio and video data
– Payload type identification
– Packet sequence numbering
– Time-stamping
• RTP runs on top of UDP (can be viewed as a
sub-layer of the transport layer)
• RTP encapsulation is only seen at the end
systems (not at the intermediate routers)
RTP - Example
• Consider sending 64 Kb/s PCM-encoded voice over RTP
• Application collects the encoded data in chunks, e.g. every
20 ms = 160 bytes/chunk
• The audio chunk, along with the RTP header, forms the
RTP packet, which is encapsulated into a UDP packet
• The RTP header indicates the type of audio encoding in
each packet. Sender can change encoding during a
conference
• RTP header also contains sequence number and
timestamps
RTP Streams
• RTP allows each source (e.g., a camera or a
microphone) to be assigned its own independent
stream of packets
– E.g., for a videoconference between two
participants, 4 streams could be opened: two
streams for audio (one in each direction) and two
for video
• However, MPEG-1 and MPEG-2 bundle audio and
video into a single stream during the encoding
process - then only one RTP stream is generated
in each direction.
Real-Time Control Protocol
(RTCP)
• Works in conjunction with RTP
• Each participant in a RTP session periodically
transmits RTCP control packets to all other
participants.
– Each RTCP packet contains sender and/or receiver
reports with statistics useful to the application
– E.g.: number of packets sent, number of packets
lost, inter-arrival jitter, etc.
• This feedback information can be used to control
performance and diagnostic purposes
– E.g. the sender may modify its transmission based
on feedback
RTCP (cont’d)
• RTCP can be used to synchronize different media
stream within a RTP session
– E.g.: videoconference where each sender generates
one RTP stream for video and one for audio
– Timestamps in RTP packets are tied to video/audio
sampling clocks (not wall-clock time, ie real time)
– Each RTCP sender-report packet contains, for the
most recently generated packet in the associated
stream, the timestamp of the RTP packet and the
wall-clock time of when the packet was created
– Receivers can use this association to synchronize the
playout of audio and video
Protocols for Real-Time (cont’d)
• RTSP (Real-Time Streaming Protocol) provides
methods to realize commands (play, fast-forward,
fast-rewind, pause, stop) similar to the functionality
provided by CD players or VCRs.
– Can control either a single or several timesynchronized streams of continuous media.
– Can act as a network remote control for
multimedia servers and can run over TCP or
UDP
Live vs. Video-On-Demand
• Live streaming is a live broadcast which allows
users to join a session in which real time media is
being sent over a network
– Because these streams are live, users are not
allowed to jump around to any point in time
– Live streaming can exploit multicast or broadcast
protocols
• Video-on-demand represents content stored on a
streaming server which can be viewed at any time
– User is allowed to jump to any point in time in the
media
– Only unicast protocols are allowed