Cs 414 multimedia systems design lecture 22 multimedia session protocols
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CS 414 – Multimedia Systems Design Lecture 22 – Multimedia Session Protocols. Klara Nahrstedt Spring 2014. Administrative. MP1,HW1, Midterm graded Pick up your HW1 and midterm from TA during office hours or by appointment Regrading: meet with TA or email TA asking for regarding

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CS 414 – Multimedia Systems Design Lecture 22 – Multimedia Session Protocols

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Cs 414 multimedia systems design lecture 22 multimedia session protocols

CS 414 – Multimedia Systems DesignLecture 22 – Multimedia Session Protocols

Klara Nahrstedt

Spring 2014

CS 414 - Spring 2014


Administrative

Administrative

  • MP1,HW1, Midterm graded

  • Pick up your HW1 and midterm from TA during office hours or by appointment

  • Regrading: meet with TA or email TA asking for regarding

    • He will facilitate the regarding with the graders

    • Deadline until Wednesday, March 19 (one week)

      • If you can’t make the deadline, email to TA and Instructor

CS 414 - Spring 2014


Outline

Outline

  • Error Control

  • Multimedia Protocols – Standards

    • RTP/UDP/IP – Transmission Protocol

    • RTCP Control/Negotiation Protocol to RTP

CS 414 - Spring 2014


Covered aspects of multimedia

Covered Aspects of Multimedia

Audio/Video

Presentation

Playback

Image/Video

Capture

Audio/Video

Perception/ Playback

Image/Video Information

Representation

Transmission

Transmission

Compression

Processing

Audio

Capture

Media

Server

Storage

Audio Information

Representation

A/V

Playback

CS 414 - Spring 2014


Error detection

Error Detection

  • Ability to detect the presence of errors caused by noise or other impairments during transmission from sender to receiver

    • Traditional mechanisms: check-summing, PDU sequencing

      • Checksum of a message is an arithmetic sum of message code words of a certain word length (e.g., byte)

      • CRC – Cyclic Redundancy Check – function that takes as input a data stream of any length and produces as output a value (commonly a 32-bit integer) – can be used as a checksum to detect accidental alteration of data during transmission or storage

    • Multimedia mechanisms: byte error detection at application PDU, time detection

CS 414 - Spring 2014


Design of error correction codes

Design of Error Correction Codes

  • Automatic repeat-request (ARQ)

    • Transmitter sends the data and also an error detection code, which the receiver uses to check for errors, and requests retransmission for erroneous data

    • The receiver sends ACK (acknowledgement of correctly received data)

  • Forward Error Correction (FEC)

    • Transmitted encodes the data with an error-correcting code (ECC) and sends the coded msg. No ACK exists.

CS 414 - Spring 2014


Error control

Error Control

  • Error Correction

    • Traditional mechanisms: retransmission using acknowledgement schemes, window-based flow control

    • Multimedia mechanisms:

      • Go-back-N Retransmission

      • Selective retransmission

      • Partially reliable streams

      • Forward error correction

      • Priority channel coding

      • Slack Automatic Repeat Request

CS 414 - Spring 2014


Go back n retransmission

Go-back-N Retransmission

CS 414 - Spring 2014


Internet multimedia protocol stack

Internet Multimedia Protocol Stack

Media encaps

(H.264, MPEG-4)

DASH

APPLICATION

RTSP

RSVP

RTCP

Layer 5

(Session)

SIP

RTP

HTTP

Layer 4

(Transport)

TCP

DCCP

UDP

KERNEL

Layer 3

(Network)

IP Version 4, IP Version 6

AAL3/4

AAL5

MPLS

Layer 2

(Link/MAC)

Ethernet/WiFi

ATM/Fiber Optics

CS 414 - Spring 2014


Internet multimedia protocol stack1

Internet Multimedia Protocol Stack

Media encaps

(H.264, MPEG-4)

DASH

APPLICATION

RTSP

RSVP

RTCP

Layer 5

(Session)

SIP

RTP

HTTP

Layer 4

(Transport)

TCP

DCCP

UDP

KERNEL

Layer 3

(Network)

IP Version 4, IP Version 6

AAL3/4

AAL5

MPLS

Layer 2

(Link/MAC)

Ethernet/WiFi

ATM/Fiber Optics

CS 414 - Spring 2014


Service requirements for real time flows voice video

Service Requirements for Real-time Flows (Voice/Video)

Sequencing

Intra-media synchronization

Inter-media synchronization

Payload identification

Frame indication

CS 414 - Spring 2014


Real time transmission protocol rtp

Real-time Transmission Protocol (RTP)

RTP

RTCP

Layer 4

User Datagram Protocol

Internet Protocol

Ethernet 802.13 or Wi-Fi 802.11

PHY (Wired or Wireless)

RTP provides end-to-end transport functions suitable for real-time audio/video applications over multicast and unicast network services

RTP companion protocol – Real-time Transport Control Protocol (RTCP)

CS 414 - Spring 2014


Relation between rtp and rtcp

Relation between RTP and RTCP

Application

Application

Decoding

Coding

Coding

Decoding

RTP

RTCP

RTCP

RTP

UDP/IP

UDP/IP

CS 414 - Spring 2014


Rtcp control and management

RTCP: Control and Management

  • Out-of-band control information for RTP flow.

    • Monitors QoS for RTP in the delivery and packaging of multimedia data

    • Used periodically to transmit control packets to participants in a streaming multimedia session.

    • Provides feedback on the quality of service being provided by RTP.

    • Gathers statisticson media connection

      • Bytes sent, packets sent, lost packets, jitter, feedback and round trip delay.

      • Application may use this information to increase the quality of service, perhaps by limiting flow or using a different codec.

CS 414 - Spring 2014


Rtcp functions

RTCP Functions

  • There are several type of RTCP packets:

    • Sender report packet,

    • Receiver report packet,

    • Source Description RTCP Packet,

    • Goodbye RTCP Packet and

    • Application Specific RTCP packets.

  • RTCP itself does not provide any flow encryption or authentication means. SRTCP protocol can be used for that purpose.

CS 414 - Spring 2014


Rtp services

RTP Services

  • Payload Type Identification

    • Determination of media coding

    • Source identification

    • RTP works with Profiles

      • Profile defines a set of payload type codes and their mappings to payload formats

  • Sequence numbering

    • Error detection

  • Time-stamping

    • Time monitoring, synchronization, jitter calculation

  • Delivery monitoring

CS 414 - Spring 2014


Rtp services support of heterogeneity

RTP Services – Support of Heterogeneity

  • Mixer service

    • Allows for resynchronization of incoming audio packets

    • Reconstructs constant 20 ms spacing generated by sender

    • Mixes reconstructed audio streams into single stream

    • Translates audio encoding to lower bandwidth

    • Forwards lower bandwidth packet streams

  • Translator service

    • Allows for translation between IP and other high speed protocols

    • May change encoding data

CS 414 - Spring 2012


Difference between mixers and translators

Difference between Mixers and Translators

S3

S1

R1

M1

M2

T

S2

S4

CS 414 - Spring 2012


Rtp common header

RTP Common Header

Ver – Version 2

P – Padding

X – Extension, if set, the fixed head is followed by exactly one

header extension

CC – CSRC count

M – Marker – intended to allow significant events such as

frame boundaries to be marked (defined by profile)

PT – Payload type

SSRS – synchronization source, CSRC – contribution source

CS 414 - Spring 2012


Payload formats

Payload Formats

  • Static Payload formats

    • Established in RTP Profile

    • Payload type 0 := µ-law audio codec

  • Dynamic Payload formats

    • Applications agree per session on payload format

    • H.263, JPEG, MPEG

CS 414 - Spring 2012


Conclusion

Conclusion

RTP usage – in several application audio and video tools (vat, vic)

RTP follows the principle of application level framing and integrated layer processing

RTP/UDP/IP is being used by the current streaming session protocols such as RTSP

Session protocols are actually negotiation/session establishment protocols that assist multimedia applications

Multimedia applications such as QuickTime, Real Player and others use them

CS 414 - Spring 2014


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