Transfer and Control Protocols Standards of ITU H.261

Transcrição

Lehrstuhl für Informatik 4
Kommunikation und verteilte Systeme
Transfer and Control Protocols
Chapter 2: Basics
Chapter 3: Multimedia Systems – Communication Aspects and Services
• Multimedia Applications and Communication
• Multimedia Transfer and
Control Protocols
• Quality of Service and
Resource Management
3.2: Transfer and Control Protocols
• Synchronization
• The H.x Protocols
• Multimedia Operating Systems
• Session Initiation Protocol SIP
Chapter 4: Multimedia Systems
• Streaming Multimedia Data
– Storage Aspects
• Transport Protocols: RTP and
Chapter 5: Multimedia Usage
RTCP
• VoIP Example
Page 1
Chapter 3.2: Transfer and Control Protocols
A main protocol family is the H.x standards by the ITU
• H.261 and H.263 define video coding for video conferences, similar to MPEG
• H.323 is a control protocol for cooperative computing (session management)
• Developed by ITU, driven by telecommunication needs
Alternative for session management: Session Initiation Protocol (SIP)
• Only one protocol, not a protocol family
• Developed by IETF: integrated with the Internet
Additionally: RTP/RTCP as transfer protocols
• H.x and SIP both are not defining transport subsystems
• RTP as an addition to UDP
Standards of ITU
The ITU has standardized everything needed
in cooperative computing:
• G.711, G.722, G.723, G.728, G.729 for audio
coding with 5.3 – 64 kbit/s
• H.261, H.263, H.264, … for video coding
similar to MPEG
• H.245 for controlling media streams
• H.450 for negotiation of communication
resources
• H.235 for authentication and ciphering
• H.225.0 for connection setup and termination,
packetizing of data streams, signaling, …
• H.323 for controlling and coordination
• … and several more, e.g. T.x for data transfer
Page 2
H.261
User Interface
Audio
Video
Audio
Codecs
G.711
G.722
…
Video
Codecs
H.261
H.263
…
Configuration
H.245
H.450
H.235
For video conferencing systems, coding/decoding in real-time is required
• H.261 was designed for ISDN
• It is a video codec for audiovisual services at p·64 Kbit/s (p = 1, 2, 3, ..., 30,
referring to ISDN)
→ H.261 can be denoted as “px64”
• Real-time processing requirement of encoding and decoding considered in this
standard: maximum signal delay ≤ 150 ms (this requirement is a kind of
limitation concerning coding and decoding procedures)
H.323
H.225.0 Layer
Network Interface
Page 3
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Properties of H.261
Image Preparation
• Image is subdivided into blocks of size 8 lines × 8 pixels (luminance & chrominance)
• Macro blocks consists of 4 luminance blocks and 2 corresponding chrominance blocks
• Group of blocks (GOB) = combination of 33 macro blocks
→ QCIF image consists of 3 GOBs (= 3 × 33 × 4 × 8 × 8 = 144 × 176 pixels for
luminance), CIF image of 12 GOBs (= 12 × 33 × 4 × 8 × 8 = 288 × 352 pixels for
luminance)
• Note: color difference samples placed such that their block boundaries coincide with
luminance block boundaries:
• Image format precisely defined
• Image refresh frequency at input: 29.97 frames/sec
• Image encoded as luminance signal Y and chrominance difference signals CB, CR
(according to a 4:1:1 subsampling scheme, later adopted by MPEG)
→ 3 basic information from which the full color may be constructed
• 2 resolution formats (each with 4:3 aspect ratio):
Common Intermediate Format (CIF)
• luminance 288 lines × 352 pixels (8 bit per pixel)
• chrominance 144 × 176
Quarter-CIF (QCIF)
• luminance 144 × 176
• chrominance 72 × 88
QCIF is mandatory for all H.261 implementations, CIF is optional
Luminance sample
Chrominance sample
Block edge
Block
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Data Amount
Interframe Coding
Uncompressed QCIF:
• Data rate = 29.97 frames/sec × (144 × 176 + 2 × 72 × 88) bytes/frame ≈ 9.115 Mbit/sec
Uncompressed CIF:
• Data rate ≈ 36.45 Mbit/sec (= 4 × 9.115 Mbit/sec)
Compressed QCIF:
• Needs only 10 frames per second (instead of 29.97 frames/sec), i.e. three times less:
3.041 Mbit/sec are required
→ Compression ratio in order to transmit 3.041 uncompressed Mbit/sec via a 64 Kbit/sec
line: 64/3.041 ≈ 1 : 47.5
This is possible for today's technology, but only for slow moving pictures
• Compressed CIF would need 4 - 6 ISDN B-channels for the same purpose
• Coding Algorithms: 2 different methods of coding (choice up to the coding control
strategy):
Intraframe coding: like in JPEG with DCT, quantization and entropy encoding
Interframe coding: use of information from previous frame (P frames in MPEG)
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Prediction for each macro block by motion compensation and spatial filter
• Motion compensation (similar to MPEG):
Comparison of macro blocks from previous and current image
→ motion vector defined by relative position of previous and current macro block
One motion vector per macro block, used for all luminance and chrominance blocks
• Simple implementations just compare previous and actual macro blocks at the same
position. In such case, the motion vector is a zero vector
• Optionally (but rarely used) a low pass filter between DCT and entropy encoding can be
used for deleting any remaining high-frequency noise
• Linear quantization (step size adjusted according to data amount in transformation buffer)
Constant data rate at encoder output enforced
Quality of encoded video data depends on image contents and motion within scene
Page 8
Data Streams in H.261
H.263 and H.264
Characteristics of Data Stream for H.261:
• Data stream produced by H.261 has a hierarchical structure
→ several layers, like in MPEG (bottom layer containing compressed picture)
• Data stream includes information for error correction (18 parity bits for 492 data bits)
• 5-bit image number as temporal reference for each image
• Freezing of image which was shown last is possible by an application command; this
allows the application at the decoding station to stop and start a video scene in a
convenient way
• Switching between still images and moving images possible (by encoder command!)
H.263 is similar to H.261, but…
• … defines 5 image formats (sub-QCIF, QCIF, CIF, 4CIF, 16CIF)
• … error correction is optional
• … consideration of GOBs and Slices like in MPEG
H.264 additionally …
• … allows variable block size (16×16, 8×16, 16×8, 8×8)
• … uses a very simple 4×4 transform instead of DCT (astonishingly with negligible loss in
quality!)
• … allows to use any frame as a reference for prediction
• … also allows bi-directional prediction (B-frames)
• … allows mixed frames – slices of one frame can be coded independently as I-slices, Pslices and B-slices!
Conclusion:
• Suited for applications which do not require too much quality and where the content
doesn‘t move too fast (video conferencing)
Page 9
H.323
The ITU Family and the OSI Reference Model
APPLICATION
A video conference is not only transferring video…
• Audio transmission (G.7xx), synchronization with video stream
• Exchange of configuration data, signalling (H.225, H.245)
• Whiteboard, chat, application sharing, data, fax … (T.x)
• Transport subsystem (TCP, UDP, RTP, RTCP)
• …
→ H.323 for coordination
G.711
Video Signal
G.728
H.261
H.263
G.722
G.729
Audio Signal
G.723.1
Data
T.127
T.126
PRESENTATION
SESSION
T.124
H.323
RTCP
RAS
RTP
TRANSPORT
Not only client terminals (telephones, video phones, NetMeeting, …) “speak” H.323,
but also other system components:
• Gatekeeper: address translation (phone numbers to IP addresses), admission
control and bandwidth management for multipoint connections, call authorization,
call signal routing
• Gateway: integration with other voice networks
• Multipoint control unit (MCU): coordinates several terminals taking part in a
conference
• Proxy: e.g. used to pass a firewall
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T.125/
T.122
Supplementary Services
H.450.3
H.235
• Transfer of multimedia data uses
UDP, transfer of control information
uses TCP
• H.323 is an “umbrella” standard
comprising all the other functionality
H.450.2
H.450.1
X.224.0
Control
UDP
H.245
H.225
TCP
NETWORK
DATA LINK
PHYSICAL
Page 11
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H.323 Network Components
H.323 Components and Signaling
H.225/RAS messages
H.225/RAS messages
H.225/Q.931 (optional)
H.225/Q.931 (optional)
Gatekeeper
H.245 messages (optional)
H.245 messages (optional)
H.225/Q.931 messages over
call signaling channel
Terminal
• H.323 terminal can be workstations as well as more specalized end systems, e.g. IP
phones
• The gateway enables an integration with existing systems like ISDN or older POTS
(Plain Old Telephony System)
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Gateway
• H.245 – A protocol for capabilities advertisement, media channel establishment and
conference control.
• H.225 - Call Control
• Q.931 – A protocol for call control and call setup.
• RAS – Registration, admission and status protocol used for communicating between
an H.323 endpoint and a gatekeeper.
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Process for Establishing Communication
Simplified H.323 Call Setup
Establishing communication using H.323 occurs in five steps:
1. Call setup
2. Initial communication and capabilities exchange
3. Audio/video communication establishment
4. Call services
5. Call termination
POTS
H.245 messages over call
control channel
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• Both endpoints have previously
registered with the gatekeeper
• Terminal A initiate the call to the
gatekeeper
• The gatekeeper provides information for
Terminal A to contact Terminal B
• Terminal A sends a SETUP message to
Terminal B
• Terminal B responds with a Call
Proceeding message and also contacts
the gatekeeper for permission
• Terminal B sends a Alerting and
Connect message
• Terminal B and A exchange H.245
messages to determine master/slave,
terminal capabilities, and open logical
channels
• The two terminals establish RTP media
paths3.2:
for Transfer
data transmission
Chapter
and Control Protocols
Terminal A
Gatekeeper
Terminal B
1. ARQ
2. ACF
3. SETUP
4. Call Proceeding
5. ARQ
6. ACF
7.Alerting
8.Connect
H.245 Messages
RTP Media Path
RAS messages
Call Signaling Messages
Note: This diagram only illustrates a simple pointto-point call setup where call signaling is not routed
to the gatekeeper. Refer to the H.323
recommendation for more call setup scenarios.
Page 16
Session Initiation Protocol SIP
SIP Services
• Defined by IETF
• SIP long-term vision
All telephone calls and video conference calls take place over the Internet
People are identified by names or e-mail addresses, rather than by phone
numbers
You can reach the callee, no matter where the callee roams, no matter what IP
device the callee is currently using
• SIP is an application layer signaling protocol that defines initiation, modification and
termination of interactive multimedia communication sessions between multiple users
• Bases upon HTTP concepts (message syntax, SIP URLs, responses, …)
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Determine current IP address of callee
• Maps mnemonic identifier to current IP address
Call management
• Add new media streams during call
• Change encoding during call
• Invite others
• Transfer and hold calls
Page 18
Setting up a Call to a known IP Address
µ
Setting up a call
• Provides mechanisms for caller to let callee know he wants to establish a call
• Provides mechanisms so that caller and callee can agree on media type and encoding
• Provides mechanisms to end call
Call Setup
• Alice’s SIP invite message
indicates her port number & IP
address. Indicates encoding
that Alice prefers to receive
(PCM µlaw)
• Bob’s 200 OK message
indicates his port number, IP
address & preferred encoding
(GSM)
• SIP messages can be sent
over TCP or UDP; here sent
over RTP/UDP
• Default SIP port number is
506.
Page 19
• Codec negotiation
Suppose Bob doesn’t have PCM µlaw encoder
Bob will instead reply with 606 Not Acceptable Reply and list encoders he
can use
Alice can then send a new INVITE message, advertising an appropriate
encoder
• Rejecting the call
Bob can reject with replies “busy,” “gone,” “payment required,” “forbidden”
• Media can be sent over RTP or some other protocol.
Page 20
Example of SIP message
Name Translation and User Location
Here we don’t know
Bob’s IP address.
Intermediate SIP
servers will be
necessary.
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 167.180.112.24
From: sip:[email protected]
To: sip:[email protected]
Call-ID: [email protected]
Content-Type: application/sdp
Content-Length: 885
Alice sends and
receives SIP messages
using the SIP default
port number 506.
c=IN IP4 167.180.112.24
m=audio 38060 RTP/AVP 0
Alice specifies in Via:
header that SIP client
sends and receives
SIP messages over UDP
Notes:
• HTTP message syntax
• sdp = session description protocol
• Call-ID is unique for every call.
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• Caller wants to call callee, but only has callee’s name or e-mail address
• Need to get IP address of callee’s current host:
User moves around
DHCP protocol
User has different IP devices (PC, PDA, car device)
• Result can be based on:
Time of day (work, home)
Caller (don’t want boss to call you at home)
Status of callee (calls sent to voicemail when callee is already talking to someone)
Service provided by SIP servers:
• SIP registrar server
• SIP proxy server
• SIP redirect server
• SIP location server
SIP Distributed Architecture
SIP Registrar
• User Agent Client
(UAC) – An entity
that initiates a call
• User Agent Server
(UAS) – An entity
that receives a call
SIP Components
Location
Server
Redirect
Server
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Registrar
Server
Register Message:
•
•
•
•
•
PSTN
User Agent
• When Bob starts SIP client, the client sends SIP REGISTER message to Bob’s
registrar server
REGISTER sip:domain.com SIP/2.0
Via: SIP/2.0/UDP 193.64.210.89
From: sip:[email protected]
To: sip:[email protected]
Expires: 3600
Gateway
Proxy
Server
Proxy
Server
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SIP Proxy
Example
SIP registrar
upenn.edu
• Alice sends invite message to her proxy server
contains address sip:[email protected]
Caller [email protected] places a
call to [email protected]
(1) Jim sends INVITE message
to umass SIP proxy
(2) Proxy forwards request to
upenn registrar server
(3) upenn server returns
redirect response,
indicating that it should
try [email protected]
(4) umass proxy sends INVITE
to eurecom registrar
(5) eurecom regristrar forwards
INVITE to 197.87.54.21,
which is running keith’s SIP
client
• Proxy responsible for routing SIP messages to callee
possibly through multiple proxies
• Callee sends response back through the same set of proxies
• Proxy returns SIP response message to Alice
contains Bob’s IP address
• Interprets, rewrites or translates a request message before forwarding it
• Note: proxy is analogous to local DNS server
Page 25
4
5
7
8
6
9
SIP client
217.123.56.89
SIP client
197.87.54.21
(6-8) SIP response sent back
(9) Messages sent directly between clients
Note: also a SIP ack message, which is not shown
Page 26
Transport Subsystem
• H.323 comes from the ITU (telephony)
• SIP comes from IETF: Borrows much of its
concepts from HTTP. SIP has a Web
flavor, whereas H.323 has a telephony
flavor
How to transfer multimedia data in the Internet?
• TCP/UDP/IP: “best-effort service”
• No guarantees on delay, loss
→ Today’s Internet multimedia applications use application-level techniques to
mitigate (as best as possible) effects of delay and loss
E.g. streamed stored multimedia
• Application-level streaming techniques for making the best out of best effort service:
Client side buffering
Use of UDP versus TCP
Multiple encodings of multimedia
But: what protocols on lower layers are suitable to support such application-level
streaming?
H.323 is complex – SIP uses the KISS principle: Keep it simple and stupid
1
3
Comparison with H.323
• H.323 is a complete, vertically
integrated suite of protocols for
multimedia conferencing: signaling,
registration, admission control,
transport and codecs
• SIP is a single component. Works
with RTP, but does not mandate it.
Can be combined with other
protocols and services.
SIP
registrar
eurecom.fr
2
SIP proxy
umass.edu
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Internet Multimedia: Simplest Approach
Internet Multimedia: Streaming Approach
First: how can application level streaming be realized?
• Audio or video stored in files
• Files are transferred as HTTP object
Received in entirety at client
Then passed to player
•
•
•
•
Audio and video are not really streamed:
• Long delays until playout!
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Browser GETs metafile with server contact information
Browser launches player, passing metafile
Player contacts server
Server streams audio/video to player
Page 30
Streaming from a Streaming Server
Streaming Multimedia: Client Buffering
constant
drain
rate d
variable fill
rate x(t)
buffered
video
• Separation of web server and streaming
• This architecture allows for non-HTTP protocol between server and media player
• Can also use UDP instead of TCP
Page 31
• In streaming, data can arrive with variable rate by network delay and jitter
• Thus: client-side buffering for playout delay for compensation of these problems
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Streaming Multimedia
Solution: RTSP
What transport protocol to use for such an approach?
HTTP
• Does not target multimedia content
• No commands for fast forward, etc.
UDP
• Server sends at rate appropriate for client (oblivious to network congestion !)
• Often send rate = encoding rate = constant rate
• Then: fill rate = constant rate - packet loss
• Short playout delay (2-5 seconds) to compensate for network jitter
• Error recovery: if time permits
Real-time Streaming Protocol RTSP
• Client-server application layer protocol
• For user to control display: rewind, fast forward, pause, resume, repositioning, etc…
What it doesn’t do:
• Does not define how audio/video is encapsulated for streaming over network
• Does not restrict how streamed media is transported; it can be transported over UDP
or TCP
• Does not specify how the media player buffers audio/video
TCP
• Send at maximum possible rate under TCP
• Fill rate fluctuates due to TCP congestion control
• Larger playout delay: smooth TCP delivery rate
• HTTP/TCP passes more easily through firewalls
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RTSP: Out of Band Control
RTSP Example
Scenario:
• Metafile communicated to web browser
• Browser launches player
• Player sets up an RTSP control connection and a data connection to streaming server
FTP uses an “out-of-band” control channel:
• A file is transferred over one TCP connection
• Control information (directory changes, file deletion, file renaming, etc.) is
sent over a separate TCP connection
• The “out-of-band” and “in-band” channels use different port numbers
RTSP messages are also sent out-of-band:
• RTSP control messages use different port numbers than the media stream
(Port 554): out-of-band
• The media stream is considered “in-band”
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Metafile Example
RTSP Exchange Example
<title>Twister</title>
<session>
<group language=en lipsync>
<switch>
<track type=audio
e="PCMU/8000/1"
src = "rtsp://audio.example.com/twister/audio.en/lofi">
<track type=audio
e="DVI4/16000/2" pt="90 DVI4/8000/1"
src="rtsp://audio.example.com/twister/audio.en/hifi">
</switch>
<track type="video/jpeg"
src="rtsp://video.example.com/twister/video">
</group>
</session>
S: RTSP/1.0 200 1 OK
Session 4231
C: PLAY rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0
Session: 4231
Range: npt=0C: PAUSE rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0
Session: 4231
Range: npt=37
C: TEARDOWN rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0
Session: 4231
S: 200 3 OK
Page 37
Example: Internet Phone
Jitter
Cumulative data
Introduce Internet Phone by way of an example
• Speaker’s audio: alternating talk spurts, silent periods
64 kbit/s during talk spurt
• Packets are generated only during talk spurts
20 msec chunks at 64 kbit/sec: 160 bytes data
• Application-layer header added to each chunk
• Chunk and header are encapsulated into a UDP segment.
• Application sends UDP segments into socket every 20 msec during talkspurt
Required:
• Network loss: IP datagram lost due to network congestion (router buffer overflow)
• Delay loss: IP datagram arrives too late for playout at receiver
Delays: processing, queueing in network; end-system (sender, receiver) delays
Typical maximum tolerable delay: 400 ms
• Loss tolerance: depending on voice encoding, losses concealed, packet loss rates
between 1% and 10% can be tolerated
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constant bit
rate
transmission
client
reception
variable
network
delay
(jitter)
client playout
delay
constant bit
rate playout
at client
buffered
data
C: SETUP rtsp://audio.example.com/twister/audio RTSP/1.0
Transport: rtp/udp; compression; port=3056; mode=PLAY
time
• Consider the end-to-end delays of two consecutive packets: difference can be
more or less than 20 msec
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Internet Phone: Fixed Playout Delay
Adaptive Playout Delay
Receiver attempts to playout each chunk exactly q msecs after chunk was generated
• chunk has timestamp t:
r: receiving of first packet
play out chunk at t + q
p: first playout schedule
• chunk arrives after t + q:
p’: second playout schedule
data arrives too late
for playout, data “lost”
Tradeoff for q:
• large q:
less packet loss
• small q: better
interactive experience
• Goal: minimize playout delay, keeping late loss rate low
• Approach: adaptive playout delay adjustment:
Estimate network delay, adjust playout delay at beginning of each talk spurt
Silent periods compressed and elongated
Chunks still played out every 20 msec during talk spurt.
ti = timestamp of the ith packet
ri = the time packet i is received by receiver
pi = the time packet i is played at receiver
di* = ri − t i = network delay for ith packet
di = estimate of average network delay after receiving ith packet
Dynamic estimate of average delay at receiver:
where u is a fixed constant
(e.g., u = .01)
20
msec
d i = (1 − u ) ⋅ d i −1 + u ⋅ d i*
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Adaptive Playout Delay
Recovery from Packet Loss
Also useful to estimate the average deviation of the delay vi (jitter):
v i = (1 − u ) ⋅ v i −1 + u⋅ | d i* − d i |
The estimates di and vi are calculated for every received packet, although they are only
used at the beginning of a talk spurt.
For first packet in talk spurt, playout time is:
pi = t i + d i + Kv i
where K is a positive constant.
Remaining packets in talkspurt are played out periodically
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Forward error correction (FEC): simple scheme
• For every group of n chunks create a redundant chunk by exclusive OR-ing the n
original chunks
• Send out n+1 chunks, increasing the bandwidth by factor 1/n.
• Can reconstruct the original n chunks if there is at most one lost chunk from the n+1
chunks
• Playout delay needs to be fixed to the time to receive all n+1 packets
• Tradeoff:
increase n, less bandwidth waste
increase n, longer playout delay
increase n, higher probability that 2 or more chunks will be lost
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Other FEC scheme:
• “Piggyback lower
quality stream”
• Send lower resolution
audio stream as the
redundant information
• For example, nominal
stream PCM at 64 kbps
and redundant stream
GSM at 13 kbps
Interleaving
• Chunks are broken up into smaller units
• For example, 45 msec units per chunk
• Packet contains small units from
different chunks
lower quality
• Whenever there is non-consecutive loss, the receiver can conceal the loss
• Can also append (n-1)st and (n-2)nd low-bit rate chunk
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Summary: Internet Multimedia: Bag of Tricks
Real-Time Protocol (RTP)
• Use UDP to avoid TCP congestion control (delays) for time-sensitive traffic
• Client-side adaptive playout delay to compensate for network delay
• Server side matches stream bandwidth to available client-to-server path bandwidth
Chose among pre-encoded stream rates
Dynamic server encoding rate
• Error recovery (on top of UDP)
FEC, interleaving
Retransmissions, time permitting
Conceal errors: repeat nearby data
→ Provide a standardized transport protocol which supports such tricks: RTP
• If packet is lost, still have most of every
chunk
• Has no redundancy overhead
• But adds to playout delay
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RTSP still would have to use the unreliable UDP or the “slow” TCP – better define a
“new” transport protocol for combining speed with reliability:
Real-Time Transport Protocol (RTP)
• RTP specifies a packet structure for packets carrying audio and video data
• RTP packet provides
Payload type identification
Packet sequence numbering
Timestamping
• RTP runs in the end systems
• RTP packets are encapsulated in UDP segments
• Interoperability: if two Internet phone applications run RTP, then they may be able to
work together
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RTP runs on Top of UDP
RTP Header
RTP libraries provide a transport-layer interface that extend UDP:
• Port numbers, IP addresses
• Payload type identification
• Packet sequence numbering
• Time-stamping
Transport
Layer
Page 49
•
•
•
•
•
•
Ver.: Version number of the RTP protocol in use
P: packet size was padded to a multiple of 32 bit
X: an extension header is used
CC: indicates the number of sources
M: User-specific mark. Can e.g. mark the beginning of a word on an audio channel.
Contributing Source Identifier: used by mixers in the studio. The mixed flows are listed here.
Page 50
RTP Header
RTP Header
Payload Type (7 bits)
Indicates type of encoding currently being used. If the sender changes encoding in
middle of transmission, it informs the receiver through this payload type field
• Payload type 0: PCM µ-law, 64 kbps
• Payload type 3, GSM, 13 kbps
• Payload type 26, Motion JPEG
• Payload type 31, H.261
• Payload type 33, MPEG2 video
Timestamp field (32 bits long)
• Reflects the sampling instant of the first byte in the RTP data packet
• For audio, timestamp clock typically increments by one for each sampling period (for
example, each 125 µsecs for a 8 KHz sampling clock)
• If application generates chunks of 160 encoded samples, then timestamp increases by
160 for each RTP packet when source is active. Timestamp clock continues to increase
at constant rate when source is inactive.
• The timestamp gives the receiver the relative time (with respect to the first data) when
to playout the data
Sequence Number (16 bits)
Increments by one for each RTP packet sent, and may be used to detect packet loss and
to restore packet sequence
Synchronization Source Identifier field (32 bits long)
• Identifies the source of the RTP stream
• Each stream in a RTP session should have a distinct identifier
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RTP and QoS
Real-Time Control Protocol (RTCP)
• RTP only adds some information to the UDP header needed for kind of reliability
• RTP does not provide any mechanism to ensure timely delivery of data or provide
other quality of service guarantees
• RTP encapsulation is only seen at the end systems: it is not seen by intermediate
routers.
Routers providing best-effort service do not make any special effort to ensure
that RTP packets arrive at the destination in a timely matter.
• Usage of (and reaction to) the information in the RTP header are left over to the
application
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• Works in conjunction with RTP
• Each participant in RTP session periodically transmits RTCP control packets to all
other participants
• Each RTCP packet contains sender and/or receiver reports
report statistics useful to application
• Statistics include number of packets sent, number of packets lost, interarrival jitter,
etc.
• Feedback can be used to control performance
Sender may modify its transmissions based on feedback
• To limit traffic, each participant reduces his RTCP traffic as the number of conference
participants increases
Page 54
RTCP
Application Example: Voice over IP (VoIP)
RTCP controls the data flow:
• Feedback to the sender about QoS on receiver side
• Data losses, delay and jitter are reported
• Note: RTCP does not provide corrective actions - this is left over to the application
Sender
Application
Receiver
Application
RTP / RTCP
RTP / RTCP
UDP
UDP
IP
IP
RTP
RTCP
RTP
RTP
RTCP
RTP
RTP
RTCP
Telephony using an IP network with standardized protocols: VoIP
• Transferring speech and
signaling information
IP network
• Not only internally in a
IP phones
(Internet/Intranet)
IP network, also
integration with “normal”
IP terminal
telephony systems
IP addresses and
virtual phone numbers
VoIP
Gateway
Telecommunication
network
RTP
RTCP
ISDN phone
Page 55
Phone numbers
Page 56
VoIP-based Telephony System
Realization with H.323
H.323 zone
Company central
Branch
ISDN
ISDN
PTSS
ISDN
PTSS
Teleworking
PCs
VoIPGateway
Network
without QoS guarantees
VoIPGateway
IP network
(Internet)
R
H.323
terminal
H.323
terminal
MCU
Gateway
POTS
R
R = Router
PTSS = Private Telecommunications Switching System
Page 57
ATM network
ISDN
H.324
Gatekeeper
H.320
H.321
H.323 gives us all functionality we need to realize an IP-based telephony integrated with
conventional solutions
Page 58
Call Setup
VoIP Future
PTSS
IP phone
Intranet
ISDN
ISDN phone
VoIP
Gateway
GK
Phone nr =>IP-Adr.?
Ringing
IP[TCP[SETUP[...] ]]
GK: Gatekeeper
Pick up
receiver
SETUP
[...]
Call
Proceeding
IP[TCP[Alerting [...]]]
Alerting
IP[TCP[Connectt[...] ]]
Connect
RTP channel
B channel
IP[UDP[RTP[Voice]]]
Voice
Call is
initiated
Dialing tone
At the moment, VoIP products based on H.323 are most popular
• But: complex, and telecommunication networks tend to converge with IP networks
→ Use protocols better integrated with the IP world
→ SIP together with a MGCP (Media Gateway Control Protocol) gets more and
more significance
→ Better integration with web applications
→ SIP seems to be the multimedia signaling protocol for the future
Connection
Still a problem: quality of an IP transmission; how to improve QoS in the Internet?
Page 59
Page 60

Transfer and Control Protocols Standards of ITU H.261

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