Streaming Media Control
Download
Report
Transcript Streaming Media Control
Streaming Media Control
The protocol components of the streaming
RTP/RTCP
RVSP
Real-Time Streaming Protocol (RTSP)
The Protocol Components of the
Streaming Media
Global multimedia data- and control architecture:
RSVP for reservation of the network resources
RTP/RTCP for transporting the real-time data and providing
QoS feedback
RTSP for controlling the streaming media delivery
SIP for initiating the multimedia sessions
SAP for announcing the streaming media sessions by
multicasting
SDP or SDF for describing the streaming media sessions
Further application level software
Introduction to RTP/RTCP
• The Real-time Transport Protocol (RTP) works
alongside with the Real-time Transport Control
Protocol (RTCP).
• In fact, RTP as a protocol is made up of 2 parts - a
data part (RTP) and a control part (RTCP).
• Together, these protocols deliver real-time traffic
with timing information and feedback on the
reception quality.
Real-time Transport Protocol (RTP)
• RTP is a transport protocol that meets the
requirements of voice and other real-time data.
• RTP can transport data continuously rather than in
bursts, and it can handle data delivery in multicast
environments.
• RTP takes care of data that TCP cannot handle.
Such data can include voice, video, etc.
• RTP was designed to run independently of the
underlying transport and network layers of the OSI
model.
Functionalities of RTP
Application
{
RTP/RTCP
Transport layer
Socket
interface
UDP
IP
Link layer
Physical layer
–
–
–
–
–
–
Timing recovery
Synchronization
Content identification
Demultiplexing
Loss detection (error-recognition)
Security services
RTP Payload
IP header
UDP header
RTP header
RTP payload
Stream Translation
Translator
Mixer
RTP Implementations
• Most implementations of RTP run in IP
environments on top of the User Datagram Protocol
(UDP).
• But RTP can also be used with other protocols such
as the Asynchronous Transfer Mode (ATM), etc.
RTP in a Multimedia context
RTP in a Multimedia Context
• The RTCP packets report on the reception quality
for a RTP session.
• The routers on the network communicate through
RSVP (Resource ReSerVation Protocol) to set aside
bandwidth for the transmission of the multimedia
data.
• The RTP header provides timing information so
that data delivery can be properly synchronized.
• The RTP header also specifies the payload type. In
this way multiple data and data compression types
are allowed.
Application of MPEG-H.261 Translator
64 kbps WAN
Legend
Translator
Local Campus
Network
MPEG-H.261
Translator
Host
Real-time Transport Control Protocol
• RTCP provides information on the quality of
reception of the data transmitted
• This information is very useful to the
applications as they can change the way they
behave on basis of RTCP reports
• The RTCP feedback information can also be a
useful tool for network managers
Traffic of RTP and RTCP Packages
Sender
Receiver
RTCP
RTCP
RTP
RTP
Internet
RTCP
RTP
Receiver
More on RTCP
• RTCP is also used to keep track of the participants in a
RTP session
• It does this by carrying the RTP source identifier,
called the canonical name (CNAME), and the
synchronization source identifier (SSRC).
• RTCP also controls the rate at which participants in a
RTP session transmit RTCP packets
• The aim of this is to keep control traffic to a
maximum of 5 percent of overall session traffic
• A RTCP BYE packet is transmitted for termination
Controlling RTCP Transmission Intervals,
Bandwidth Fitting
The control traffic is limited to the maximum 5 % of the whole
traffic by the modification of the RTCP package sending rate
The available bandwidth of the RTCP is allocated to senders and
rs 25%
receivers, the part of the senders:
all receivers together:
1 rs
Ns
At sender: sending interval of RTCP packages:Ts p
B 0.05 rs
number of senders: N s
bandwidth of the whole session: B
average RTCP package size: p
At receiver the sending interval
Nr
RTCP packages: Tr p
B 0.05 (1 rs )
number of receivers: N r
Estimation of Packet Loss
The known expression for the TCP traffic can be used for the
estimating the l packet loss ration of other traffic which uses
2
the same path
pc
l
B1 RTT
Let c=1.22, RTT=200 ms, the encoding method is DVI (ADPCM
version), 40 ms data packet, RTP/UDP/IP header 40 bytes,
codec state 4 bytes, media data 160 bytes, thus p=204 bytes,
sending rate 25 packet/s,
thus the bandwidth required by the stream:
packet
byte
byte
B1 25
204
5100
s
packet
s
From these the packet loss ratio: l = 6.0 %
Conclusions
• RTP and RTCP work together
• Together, these protocols deliver real-time
traffic with timing information and feedback on
the reception quality
Real-Time Streaming Protocol (RTSP)
Product of the Progressive Networks and the Netscape
Communications
Control protocol, one or more, for creating and controlling
synchronized continuous media streams
Supports unicast and multicast
It can operate on the top of the RTP
Goals of the RTSP
– Controlling data sending sessions in different applications, such as
media-on-demand (MoD)
– Rough synchronization of streams
» The fine tuning is on RTP level by using RTP messages
– Virtual presentations, which mean synchronized playing out from
remote servers
Protocol Stack Including RTSP
Applications
Protocols for improving the reliability of the delivery
RTSP
RTP/RTCP
TCP
UDP
IP és IGMP
Data link and Physical Layer
HTTP GET
Session/presentation desription
RTSP
Operation
web server
RTSP SETUP
client
(browser
software)
RTSP PLAY
RTP audio
RTP video
RTCP
RTSP PAUSE
RTSP TEARDOWN
media
server
The RTSP commands
Név
Description
OPTIONS
receives available options
SETUP
creates a transport
ANNOUNCE
changes the description of a media object
DESCRIBE
receives the low level description of the media object
PLAY
starts playout
RECORD
starts recoding
REDIRECT
redirects client to new server
PAUSE
pauses delivery, but holding state
SET_PARAMETER
controls tool or coder
TEARDOWN
tears the state down
Usage of the RTSP
The next addressing is for the whole presentation
rtsp://turul.eet.bme.hu:554/gamefilm
An audio track inside a presentation:
rtsp://turul.eet.bme.hu:554/gamefilm/audiotrack
The RTSP name-hierarchy is different from the URL namehierarchy used by the HTTP
Example for building RTSP into HTML
This is a part of a web-side, which contains program-guide
The role of the RTSP is setting and controlling the delivery
<session>
<group>
<track src="rtsp://audio.tv.hu/movie">
<track src="rtsp://video.tv.hu/movie">
</group>
</session>