Академический Документы
Профессиональный Документы
Культура Документы
Colin Perkins
USC Information Sciences Institute
Internet Multimedia
45
40
Packet Loss Rate (percent)
35
30
25
20
15
10
0
8:00 10:00 12:00 14:00 16:00 18:00
Time of day
Discontinuity due
to route change.
Reception time
Transmission time
Original
Fragments
Packet loss
Reconstructed
• Retransmission ensures
that no data is lost
– Reliable, in-order, delivery
Time
Slow start Slow start Congestion Congestion
avoidance avoidance
Reliable Unreliable
• RFC 2326
• RFC 2974
• The challenge:
– build a mechanism for robust, real-time media delivery
above an unreliable and unpredictable transport layer
– without changing the transport layer
• Consequences:
– Intelligence tends to "bubble-up" the protocol stack to the
end points
– The intermediate systems can be simple, and need not be
robust
• They can simply discard data they cannot deliver, since it will
be recovered end-to-end
Payload
Payload
Payload
Payload
Format RTP Profile
Format
Format
Format
UDP
IP
Payload
Payload
Payload
Payload
Format
Format
Format
Format
RTP Profile
V PX CC M PT Sequence Number
Timestamp
Synchronization source (SSRC) identifier
Payload data
Padding
Padding
• Allows RTP to support mixers
and translators
– Mixers combine several flows
into one
• E.g. Conferencing MCU
– Translators change the
format of a flow, or gateway
between different networks
• Transcode to a lower bit-rate
• Gateway between unicast
and multicast
Point-to-point
communication
via unicast
Four participants
communicating via
a multicast group
Translated: multicast to
unicast. Two participants
communicating via a
multicast group, with a
Replicated unicast: a third linked to the session
group of three using an by an RTP translator.
RTP translator/mixer to
mediate communications.
Copyright © 2002 Colin Perkins
Media Identification
Padding
Padding
• No requirements on stability
or accuracy of clock
– Implies receiver adaptation
• Many uses:
– Loss rate can be used to select amount of FEC to employ
– Jitter gives estimate of playout buffer delay at receiver
Sender Router
Internet
Network induces
timing jitter into
the media stream
Router Receiver
Transmission
Jitter affects inter-packet timing
Network
Transit
Reception
Playout
Buffer
Network transit delay
Playout
Packet discarded
due to late arrival
Playout buffering delay added
to compensate for jitter.
Reliable Unreliable
Group
Report every Report many Just regular size
2 relevant event of the events RTCP packets
immediately but not all
1 2 3 4 Original Stream
1 2 3 4 Reconstructed stream
Compress Packetize
FEC
Bit stream A 1 1 1 0 0 0 1 1 1 1 0 0 0 1
• Not all data in the packets • Some links have high bit
is equally important error rate
– Headers and codec state – Causes packet corruption
updates are vital – Detected by UDP checksum
– Media data is of variable – Packet discarded
importance
• Seriously impacts wireless
• Data used for predication
• Data used in a single frame
link performance
– Cellular, especially
Partial Checksum
at the UDP level
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 Original stream
1 5 9 13 2 6 10 14 3 7 11 15 4 8 12 16 Interleaved stream
1 5 9 13 2 6 10 14 4 8 12 16 Packet loss
1 2 4 5 6 8 9 10 12 13 14 16 Reconstructed stream
Packets delivered
service.
– No admission control
– The network accepts all
packets and tries to deliver
them.
Packets sent
• However, no guarantee of
delivery provided
– Excess packets discarded if
links become congested.
Packets delivered
– Possible rate changes
depend on the codec
– Complex feedback loop
between codec and network
Packets sent
• For RTP, implies senders
should observe receiver
feedback
– If loss fraction is non-zero,
consider sending less
– As loss decreases, consider
sending faster
• RTP provides:
– Robust, flexible and extensible media transport
– Range of error correction schemes
– Range of security solutions
• Limitations:
– Congestion control
Colin Perkins
http://www.east.isi.edu/~csp/