Streaming Media: Architecture and Protocols

Media Streaming
Architecture and Protocols
Rüdiger Gunreben
Markus Wichmann
What is streaming media?
• Streaming media allows the user to begin viewing
audio/video clips without first completely downloading
the entire file.
• After a brief initializing and buffering, the file begins
to stream.
• Most streaming media require the user to install a
downloaded player on his computer
Legend
Router
Server
Workstation
ATM Switch
Layer 2 Frame Switch
Layer 2 High Speed Switch
Integrated Layer 2/Layer 3 Switch
Network Architecture
AVI file
MPEG file
Record
Encoder
A/V
Capture
MainFrame/Cluster
Customer/User
Network
Tasks
Encoding incoming media stream(s)
• How they will do this? (Hardware, Software)
• What codec will be used?
Handles incoming requests for transmission
• Performance issues
• Structure of the system (scalability)
Routing the transmission to the customer
• Questions about speed and special technologies
Watching the transmission
• Who will benefit?
Industry overview
Format
Product
Real
Windows
Media
QuickTime
MP3
MediaCleaner
•QT SS
•Darwin
•Kasenna
MediaBase
•Icecast
•Shoutcast
Encoder
•RealProducer
•MediaCleaner
•WMEncoder
•MediaCleaner
•Sorenson
Broadcaster
•MediaCleaner
•Lame
Player
•Realplayer
•WMPlayer
•QTPlayer
•WinAmp
•Sonique
Server
•RealServer
•Kasenna
MediaBase
OSI vs. TCP/IP
• OSI: conceptually define services, interfaces, protocols
– more layers → less efficient
• Internet: provide a successful implementation
– we don’t need so many layers in practice
Application
Presentation
Application
Session
Transport
Transport
Network
Internet
Datalink
Physical
Host-tonetwork
OSI
TCP/IP
Netmeeting
WMP
TCP
UDP
IP
LAN
Packet
radio
Streaming Versus Web Server
Requirement
Web
Streaming
On-Demand
Yes
Yes
Live Streaming
No
Yes
Unicast
Yes
Yes
Multicast
No
Yes
Bandwidth Throttling
No
Yes
Content Control
No
Yes
Commercial aspects of Streaming
Media: Who makes all the money?
• The value-added chain in Streaming Media
–
–
–
–
Content Creation ( recording systems, camera/mic operators)
Content Preparation ( encoding, cutters, sysops, hardware manufact.)
Content Storage ( Storage systems manufacturers, Storage providers)
Content Delivery ( Network components, Service providers, media
players, system manufacturers)
Commercial aspects of Streaming Media:
Cisco IP/TV : pricing example
20 user sw license
client software
Wincast encoder
Mainboard
Processor
IDE HDD
64MB Parity RAM
10/100 NIC
Rack case/rails
Floppy
CDROM
Operating System
total:
$6,500
no charge
$100
$146.95
$125.95
$165
$200
$30
$495
$23.95
$67
$180
$8054,85
Commercial aspects of Streaming Media:
IBM Videocharger: pricing example
Server software
Client software
MPEG encoder
Mainboard
Processor
IDE HDD
128MB Parity RAM
10/100 NIC
Rack case/rails
Floppy
CDROM
Server OS
total:
$1,800
no charge
$3,000
$146.95
$125.95
$165
$400
$30
$495
$23.95
$67
$200
$6303,85
Commercial aspects of Streaming Media:
Pricing example for storage and delivery
• Company: Detroit Encoding, Inc.
– Low-end example:
• Disk Space: 15 MB
• 500 MB monthly transfer
• $ 10 per month
High-end example:
Disk Space: 1000 GB
No Data Transfer Limit
$ 18000 per month
• Wooow, well... very high costs...
• But: Who or what guarantees
– the smooth arrival of the data at the customer‘s site?
– the convenient usability of the data by the customer?
Several Protocols and Technologies need to work together
The main aim: real-time
• DIN 44300: „Echtzeitbetrieb ist ein Betrieb eines
Rechensystems, bei dem Programme zur Verarbeitung
anfallender Daten ständig derart betriebsbereit sind, dass
die Verarbeitungsergebnisse innerhalb einer vorgegebenen
Zeitspanne verfügbar sind.“
• Which technical implications does this definition have?
• How can we reach this aim?
• The keyword here is Quality of Service (QoS)
Criteria for QoS with Streaming
• Data Throughput
– performance of a link (in bits per second)
• Packet latency
– total delay of a packet while travelling from sender to receiver
• Packet Loss Ratio
– the ratio of the number of packets lost to the total number of packets
that have been sent
• Jitter (Packet Delay Variation)
– the sender send this:
but the receiver gets this:
Approaching QoS
• Increase bandwidth to infinity (rarely possible)
• Define Service Classes
– On the data link layer, e.g. ATM
ATM Service Category
Typical use
uncompressed voice and video
Constant Bit Rate (CBR)
VBR
Real-Time Variable Bit Rate
(rt-VBR)
Non-Real-Time Variable Bit Rate
(nrt-VBR)
Available Bit Rate (ABR)
Unspecified Bit Rate (UBR)
reserves a constant amount of bandwidth; user may
send at the negotiated bitrate any time for any
duration
compressed voice and video
(requiring small jitter)
PCR Peak Cell Rate, SCR Sustainable Cell Rate,
MBS Maximum Burst Size
(same as rt-VBR, but with no upper delay limit)
file transfers, e-mail
guaranteed minimum bandwidth (MCR), no loss
(same as ABR, but with possible loss, without MCR)
Approaching QoS
• Define suitable protocols and techniques
– and combine them, where possible,
with solutions on the data link layer
• Key Technologies:
– Multicasting
– IPv6: A new and better version of IP
– RTP, RTCP, RTSP (for video with audio)
– H.323, SIP (for Voice-over-IP)
Unicast/Multicast
128.146.199.0/24
128.146.222.0/24
128.146.116.0/24
128.146.226.0/24
Unicast
128.146.199.0/24
Sender
128.146.222.0/24
Receiver
128.146.116.0/24
Receiver
128.146.226.0/24
Receivers
Unicast
• With 4 receivers, sender must replicate the stream 4
times.
• Consider good quality audio/video streams are
about 1.5Mbps (a T1 link)
• Each additional receiver requires another 1.5Mbps
of capacity on the sender network
• Multiple duplicate streams over expensive WAN
links
Multicast
128.146.199.0/24
Sender
128.146.222.0/24
Receiver
128.146.116.0/24
Receiver
128.146.226.0/24
Receivers
Multicast
• Source transmits one stream of data for n receivers
• Replication happens inside routers and switches
• WAN links only need one copy of the data, not n
copies.
Multicast - Overview
• Unicast is one to one communications, multicast is
potentially many-to-many.
• IP multicast uses class D addresses for destination,
unicast address for source (224.0.0.0 through
239.255.255.255)
• Source (S) sends data to a group (G) which
potentially has receivers (R)
IPv6 – Next Generation IP
A new version of IP? What for?
(1 of 4)
• Address space of IPv4 will soon be exhausted
– IPv4: Only 42 billion theoretically possible addresses
– Many intranet users share one internet IP address
(see network address translation (NAT))
– Dynamic assignment of addresses on ISP-dial-in
– Population growth: 10 billion people in the year 2020
Forecast: Exhaustion of address space in 2005
A new version of IP? What for?
(2 of 4)
• Many end-to-end services cannot be used without
restrictions
A
NAT
Internet
Internet
NAT
B
A new version of IP? What for?
(3 of 4)
• Growing routing tables
– Each class A/B/C address requires a table entry
– IPv4 does‘t allow hierarchical routing
Performance loss
• Many fields of IPv4 headers hardly used
– Routers do no longer analyse those fields
– Analysis is time-consuming
Performance loss
A new version of IP? What for?
(4 of 4)
• Administrative costs with IPv4
– Manual configuration necessary
• in user terminals (here: multimedia computers)
• and/or in DHCP servers
– Increase of the number of mobile devices aggravates
this problem
• Security of data in IPv4 in the protocol itself
– Authentication and encryption not possible without
IPSec (=additional effort for configuration)
Requirements for Streaming via IPv6
• Performance
– Optimize routing
• Support administrators in configuration
(„Auto-Configuration“)
• Enlarge address space
• Simplify multicasting
• Integrate mechanisms securing data
The IPv6 header
•
•
•
•
•
•
Version: indicates IP version (binary 0110 = decimal 6)
TC = Traffic Classifier, allows priorization
Flow Label for optimized routing
Payload Length
Next Header indicates type of (optional) following header
Hop Limit is similar to Time-to-Live (TTL) in IPv4
IPv6 header vs. IPv4 header
• No IPv4 Internet Header Length
– IPv6 header always of same length
Routing performance
• No IPv4 header checksum (!)
– Higher layers check exhaustively
Routing performance
• Fragmentation mechanisms dislocated into 0ptions header
– Reason: Path MTU Discovery renders fragmentation obsolete
• IPv4 „Protocol Type“ replaced by IPv6 „Next Header“
• IPv6: „Payload Length“ really comprises only the payload, and not the
header as well (like in IPv4)
Types of IPv6 addresses
• Unicast: unique address of a single interface
• Multicast: all interfaces of a defined group
– Avoids transmission of identical data
• Anycast: the next interface of a defined group
– several terminals have a common address
– e.g. replicated webservers with same IP address each
– Machines that want to beling to an Anycast address need to
„propagate“ this information to the routers of their subnet
(using ICMP, see later on)
• NO Broadcast any longer
(displaced by Multicast and Anycast)
Advantages of IPv6 address structure
• Decentralized administration of IP network addresses
– IANA assigns TLA IDs to lower instances
– Those assign NLA IDs to lower instances again
– Those again assign the SLA IDs to organizations and companies
• These address prefixes are propagated by the routers
– Routers convey (new) prefixes to terminals/routers
– If no router is reachable the MAC address of the NIC is used to form a
link-local IP address
• Advantages: Hierarchical fast routing, fast configuration
Scopes of IPv6 addresses
link-local
• With no TLA, NLA, and SLA IDs assigned an address
is only valid on the same network link
Scopes of IPv6 addresses
link-local
site-local
• If company routers propagate an SLA ID to the end
devices then so-called site-local addresses are
formed
Scopes of IPv6 addresses
link-local
site-local
global
• With all IDs assigned a packet originating from an
IPv6 device can be routed anywhere on the globe
Multicast Addresses in IPv6
• There are pre-defined and user-defined multicast addresses
• Pre-defined e.g.:
– all routers
– all DHCP servers
• User-defined e.g.:
– all users taking part in a video conference
– all users watching a TV broadcast over the internet
• Subscription and Un-Subscription to/from user-defined
multicast sources:
– Host sends message (ICMPv6 message) to its next router
– Router adjusts its internal tables according to the requested multicast
broadcast and will pass packets with the specified multicast address
to the network link with the requesting host
– If necessary the router propagates the demands of the host to higher
routers
Data Security with IPv6
• based on IPSec
• IPSec is an integral component of IPv6,
has been optional in IPv4
• uses the IPv6 extension headers
– Authentication Header (AH)
– Encapsulated Security Payload Header (ESP)
IPv6 supporting Mobile Streaming
Windows Media
Live Content
UNICAST,
MULITCAST
License Server
Live Feed
End Users:
Home,
Business
Encoding
Workstation
On-demand Content
Windows Media
Services Server
Stored
Content
Editing
Workstation
Authoring
Streaming from a
Web Server
Streaming from a
Download & Play
Web Server
WM Server
Distribution
Playback
Windows Media 9 Series
Component Interweavement
On-Line Services
Authoring
Movie
Maker
Producer
Third
Party
Tools
WM
Encoder
& OCX
DRM License
Creation
Content
Discovery/M
etadata
Playback
Windows
Media Player
DRM License Acquisition
WMP.DLL
Server Services
DShow
WM Format SDK
WM Server
MMC Admin
WM Server
Web Admin
Codecs
DRM
DShow
WM Format SDK
WMS Admin Object
File
Networking
IIS
Windows Media
Server
Windows
Media
Player
OCX
File
Networking
DRM
Codecs
WM Codec Parameters
•
Supports streamed / local playback
•
WM comprises
–
–
–
•
ASF (WMV / WMA)
Codecs
DRM
GOAL
–
–
–
Smallest size for quality (e.g. 50% of MP3)
Near DVD quality at only 500 kbps
Maintain consistent quality
Video Smoothing (Frame Interpolation)
• Generates missing frames at playback time using opticalflow analysis
• Improves perceived video quality (motion smoothness) at
lower data rates
• Can be used to improve quality or save bandwidth by
encoding fewer frames
– Bandwidth reduction reduces operating cost for ICPs
– E.g. MSNBC encodes only 15 frames/sec, but user of WM9 Player
will experience full frame rate
WMA Professional 9: MS’ Promises
• New audio codec to handle greater-than-CD
resolution/channels
– Up to 96Khz sampling rate, 24-bits, and 8 channels (“7.1”)
• CD is stereo at 44Khz using 16-bit samples
– Targets are multi-channel music distribution and movie sound
tracks at broadband rates
• More efficient than existing technologies:
– Twice as good as Dolby Digital (AC-3) and DTS used in DVDs
– Blind listening test shows WMA Pro at 768Kbps outperforms all
existing codecs in the market
Windows Media Format
•
Advantages
–
Extensible file format
•
•
–
Support advanced MS and 3rd party codecs
Support rich media (see e.g. TeRM)
Author once, playback anywhere
•
•
Streams 28.8Kbps 1 Mbps broadband
Replay PCs, PDAs, Consumer Elec.
– Car stereos, Cellphones, In home
devices, PDAs
Windows Media Format
File Container
•
Supports files as large as 17 million terabytes
(17 000 000 000 000 000 000 bytes)
•
•
Supports multi-bitrate audio
Stores media and metadata in one file (see current TeRM version)
–
Metadata
•
•
–
Title, copyright, author, markers, script commands, etc.
ID3 v1, v2 music meta-information
Media
•
Audio, Video, rich media and Script Commands (URLs, CC)
AudioStream
Video Stream #1
Video Stream
#nRich Stream
Script Stream
Timeline
Windows Media Format
File Container
•
•
•
•
Able to select a stream, based on available bandwidth
Interleaved data optimal for request/response protocol
like HTTP
Timeline based synchronization
Bursty data can be smoothed
Feature Set
•
•
•
•
•
Timecode with Frame accurate seeking support
Live DRM
Video Smoothing (Frame Interpolation)
Rich Media Streaming (see TeRM)
VBR streaming/progressive down-load optimizations
– the higher the bandwidth the more details
•
MBR audio and multiple resolution MBR Video
(Multiple Bitrate)
Digital Rights Management
•
Prevents unauthorized distribution and protects
content owners’ rights
•
Issue/Acquire/Enforce/Manage rights
•
Allows copyright owners to encrypt ASF content
– Users must acquire “license” to play
– Compressed content cannot easily be
e-mailed, transferred, or copied with
associated license
•
Works with Windows Media codecs
•
Support for DRM v7
Codecs
•
Video:
– Microsoft MPEG-4 v2, v3
– Microsoft MPEG-4 ISO v1
– Windows Media Video v7, v8
•
Audio:
– WMA v1, v2, v7, v8
– MP3
Streaming Protocols
•
Multicast
–
–
•
Microsoft Media Stream (MMS)
–
–
–
–
•
TCP connection for commands between client and server
UDP or TCP for media content
Automatic protocol roll-over (UDP, TCP)
Can vary the packet size as needed
HTTP
–
–
•
Each packet is broadcast to multiple clients
Requires use of Windows Media Station (previously NetShow
Channel) .NSC files
Will automatically come through corporate firewalls
Will use Internet Explorer’s proxy-settings, or will use own
settings
File-based (local or network redirector)
The Realtime Protocol Suite
• RFC 1889:
– RTP, Realtime Transport Protocol:
A Transport Protocol for Real-Time Applications
• end-to-end transport of
– audio,
– video, or
– simulation data
• multicast or unicast
• does not guarantee QoS
– RTCP (Realtime control protocol)
• monitor the quality of service of RTP
• RFC 2326:
– RTSP (Realtime Streaming Protocol)
• »network remote control« of RTP streams
RTP and RTCP: Companions
1st udp port
RTP
Media Transport
receiver
sender
Quality Feedback, Control
2nd udp port
RTCP
RTP
• Why Not Use TCP?
– No need for 100% reliability
– built-in reliability makes TCP protocol slow
– Retransmission delay
TCP not suitable for Real-Time applications
UDP and IP (with multicasting) is more suitable
Retrospection:
Criteria for QoS with Streaming
• Data Throughput
– performance of a link (in bits per second)
• Packet latency
– total delay of a packet while travelling from sender to receiver
• Packet Loss Ratio
– the ratio of the number of packets lost to the total number of packets
that have been sent
• Jitter (Packet Delay Variation):
RTP over UDP/IP: Better than TCP/IP
• UDP: connectionless higher Data Throughput
• UDP: No unnecessary reliability
higher Data Throughput
but also possible packet loss
• RTP: Timestamp
a too old packet is discarded
jitter correction
• RTP: Sequence Number
correct order, independent from time of arrival,
detect packet loss
M=marker bit
indicates begin of frame
Payload Type:
mostly audio/video
encoding method
Timestamp: sampling
instant of first data octet
SSRC: sync source, random
32 bit identifier
CSRC: list of up tp
15 contributing sources (for
mixing)
RTCP: Purpose
• provide feedback on quality of data distribution
– allows sender to detect network congestion
– allows sender to detect packet loss
– allows ISP to detect network problems (when multicasting)
• functional parts within header (excerpt):
– Receiver report (RR) (sent for each SSRC)
• Loss rate sender can adapt bandwidth of stream
• jitter
• roundtrip delay
– Explicit leave (BYE)
• important when mulitcasting
RTSP: Real-Time Streaming Protocol
• acts as a “remote control” for multimedia servers
– provides commands like
•
•
•
•
•
•
•
PLAY
PAUSE
INVITE (e.g. a server can be invited into a video conference)
RECORD (e.g. in video conferences)
GET_PARAMETER (e.g. server query of packet loss to client)
REDIRECT (server asks client to use different server load balancing)
OPTIONS (e.g. acceptable data encodings)
RTSP: Real-Time Streaming Protocol
• Request/Response protocol similar to HTTP
– some of the major differences:
• out-of-band signalling (see next sheet)
• client AND server may send requests (e.g. GET_PARAMETER)
• protocol is stateful (current state of a session is saved by server)
– Reason: Developers new to RTSP able to
adapt from their existing HTTP knowledge
RTSP: Out-of-Band Signalling
• The payload is carried using a different protocol
than RTSP (mostly RTP)
• Example for an RTSP setup request:
C
S: SETUP rtsp://www.hdm-radio.de:554/kueken RTSP/1.0
Transport: RTP;unicast;client_port=5564-5565
S
C: RTSP/1.0 200 OK
Date: 5 Dec 2002 10:45 GMT
Session: 47110815
Transport: RTP;unicast;
client_port= 5564-5565;server_port=6456-6457
Voice and Video over IP:
Protocols
VoIP – Typical Configuration
VoIP Typical Configuration
Public
Switch
(LEC)
Com puter
Public
Switch
(LEC)
Local loop
Local loop
Com puter
Telephone
Telphone
TI
Line
TI
Line
H.323
ISP
Gateways
Routers
H.323
Internet
ISP
Gateways
Routers
Protocols
• Data Transport :
– RTP
• Signalling:
– IETF SIP protocol suit
– ITU-T H.323 protocol suit
• Quality of Service:
– RSVP
VoIP characteristics
• Mostly concerns Transport and Session
(application) Layer functionality
– Assuming that the underlying network provides the
required service
• Real-Time Application
Protocol stack for IP Telephony
SIP: Session Initiation Protocol
• Signalling protocol for establishing sessions
(voice, video, game, chat, calls)
• ASCII text-based messaging
• SIP messages can be transported over any protocol
(UDP, TCP, …)
• Data Transport is left to RTP
SIP protocol
• Text based (HTTP) Request/Response messages
• Methods (Requests)
• INVITE
Call user (uses SDP for session description)
• ACK
Confirm connection
• OPTIONS
Capability info exchange
• BYE
Tear down call
• CANCEL
Cancel a previous call (before completion)
• REGISTER Sign up with server, for address lookup , etc.
SIP operation in proxy mode
RTP Sessions are setup after SIP session setup is confirmed
SIP Operation: redirect mode
SIP Examples (for reference only)
INVITE sip:bob@one.example.com SIP/2.0
Via: SIP/2.0/UDP
sip.example.com;branch=7c337f30d7ce.
1;maddr=239.128.16.254;ttl=16
Via: SIP/2.0/UDP mouse.wonderland.com
From: Alice
<sip:alice@wonderland.com>;tag=1
To: Bob <sip:bob@example.com>
Call-ID:
602214199@mouse.wonderland.com
CSeq: 1 INVITE
Contact: Alice
<sip:alice@mouse.wonderland.com>
Subject: SIP will be discussed, too
H.323 Architecture
• H.323 specifies the components, protocols, and procedures
providing multimedia communication over packet-based
networks (like IP)
• Components:
– Terminal
– Gatekeeper: the focal point for all calls within the H.323
network
• Address translation, Setup/Teardown of connections, access control,
bandwidth management
– Gateway: connects two dissimilar networks
– Multipoint Control Unit: provides support for conferences
of multiple terminals
H.323 Protocols
• H.323 terminals must support the following:
– H.245 for exchanging terminal capabilities and creation
of media channels
– H.225 for call signaling and call setup
– RAS (registration, admission, status) for registration and
other admission control with a gatekeeper
– RTP/RTCP bringing audio/video packets into correct
order
– G.711 audio CODEC.
• H.323 terminals can optionally support the following:
– Video CODECs (e.g. H.263, T.120 data-conferencing
protocols,…
H.323 Call Stages
Discovery and Registration (RAS)
Call Setup (RAS/H.225/Q.931)
Call Negotiation (H.245)
Media Channel setup (H.245)
Media Transport (RTP/RTCP)
Call termination (H.245/H.225/RAS)
“Who am I?”
“Who do I want to call?”
“These are my capabilities”
”Open audio channel”
“Send audio datagrams”
“Hang up phone!”
H.323 protocol suite
SIP vs. H.323
• Both use RTP/RTCP for Data Transport
SIP vs. H.323: dissimilarities
• Which one to use for IP Telephony ?
– SIP was designed by IP guys
• Make it as simple as possible, but lacks some features
• Good for low-cost simple devices
– H.323 was designed by Telephony guys
• Make it as complete as possible Complex, powerful, more features
• Good for more serious applications
• What did industry choose?
– 3GPP* has chosen SIP as the only VoIP protocol for UMTS release 2000
– Siemens: for Voice and Multimedia over IP, H.323 is a better choice
*3GPP = 3rd generation partnership project
Conclusion for VoIP
• For IP Telephony we need a set of protocols
– For data transport: RTP
– For signaling
• IETF suggests: SIP, RTSP, SDP …
• ITU-T suggests: H.323 protocol suit
• Our vision: Most of the current solutions are
proprietary to a great extent, but will eventually
converge to standard based solution
Overall Conclusion
• Streaming is
– very expensive
– very complex
• Protocols are not always well-defined (see RTP)
• Commercialisation lags behind potential market
– Multicasting not easily usable, no actual easily
handable standards for VoIP
• Bandwidths are increasing higher quality
• Converging technologies will make things easier
• Demands are increasing, and so will the money ☺
Resources
http://www.tv-plattform.de/pdf/Symposien/B4b3.pdf
http://radio.irt.de/vida/Docs/PUBL-Ruhnke.pdf
http://www.cisco.com/warp/public/cc/pd/cxsr/ces/prodlit/wmtv4_wp.pdf
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http://www.leitch.com/custserv/wplib.nsf/Nlookup/NT000008F6
http://www.medialab.sonera.fi/workspace/Streaming_Media_Platform.pdf
http://www-svca.mercuryinteractive.com/pdf/products/whitepapers/streamingwp.pdf
http://support.instawatch.com/streamingmediaPrimer.pdf
http://geocities.com/majormms/
http://alpha.sec.nl/vip/vip_d3_1-v110.pdf
http://www.palomar.edu/at/mediaclass/Adding%20Streaming%20Media.pdf
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http://525.fims.uwo.ca/craven/525str.htm
http://service.real.com/help/library/guides/production/htmfiles/intro.htm
http://www.cs.unc.edu/hays/INLS191/lessons/digital_media_long.ppt
http://www.ibr.cs.tu-bs.de/lehre/ss02/skm/t05_txt.pdf
http://www.zottel.de/download/paper/OberthuerStudienarbeit.pdf
http://www.strongsec.com/zhw/PA/PA2_Sna06_2001.pdf
http://wwwbode.cs.tum.edu/luksch/archiv/Skripten/S2001/GridComputing/QoS.pdf
http://www.decoit.de/whitepapers/DECOIT-QOS0107.pdf
http://www.rvs.uni-hannover.de/people/boeker/vortraege/rtsp.pdf
http://mediasrv.cs.uni-dortmund.de/Lehre/SS2002/EINI_II_SS2002/pdf/Roelz-rtsp.pdf
http://www-lehre.informatik.uni-osnabrueck.de/aerpenbe/papers/real.pdf
http://lrb.cs.uni-dortmund.de/Lehre/Desktop_Video_SS1999/PDF/DT-Video.4.Verwendungen.pdf
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