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    Nepal Telecom Exam Preparation (Level 7) : Dipak Kumar Nidhi

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    Dipak Kumar Nidhi
    This document provides an overview of Real-Time Transport Protocol (RTP). It discusses why UDP rather than TCP is used to carry audio streaming, and how RTP addresses the requirements of real-time applications by providing mechanisms for stream description, stream control, session handling, and media transport. Key aspects of RTP covered include its use of UDP, sequence numbering, time-stamping, source identification, and ability to support both unicast and multicast streaming.

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    Nepal Telecom Exam Preparation (Level 7) : Dipak Kumar Nidhi

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    Dipak Kumar Nidhi
    45 views31 pages
    This document provides an overview of Real-Time Transport Protocol (RTP). It discusses why UDP rather than TCP is used to carry audio streaming, and how RTP addresses the requirements of real-time applications by providing mechanisms for stream description, stream control, session handling, and media transport. Key aspects of RTP covered include its use of UDP, sequence numbering, time-stamping, source identification, and ability to support both unicast and multicast streaming.

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    Real Time Protocol

    Original Title

    RTP

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    This document provides an overview of Real-Time Transport Protocol (RTP). It discusses why UDP rather than TCP is used to carry audio streaming, and how RTP addresses the requirements of real-time applications by providing mechanisms for stream description, stream control, session handling, and media transport. Key aspects of RTP covered include its use of UDP, sequence numbering, time-stamping, source identification, and ability to support both unicast and multicast streaming.

    Copyright:

    © All Rights Reserved
    Download as PPTX, PDF, TXT or read online from Scribd
    Download as pptx, pdf, or txt
    45 views31 pages

    Nepal Telecom Exam Preparation (Level 7) : Dipak Kumar Nidhi

    Uploaded by

    Dipak Kumar Nidhi
    This document provides an overview of Real-Time Transport Protocol (RTP). It discusses why UDP rather than TCP is used to carry audio streaming, and how RTP addresses the requirements of real-time applications by providing mechanisms for stream description, stream control, session handling, and media transport. Key aspects of RTP covered include its use of UDP, sequence numbering, time-stamping, source identification, and ability to support both unicast and multicast streaming.

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    © All Rights Reserved
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    of 31

    Nepal Telecom Exam Preparation

    (Level 7)
    Dipak Kumar Nidhi
    Real Time Protocol
    ( RTP – Real-Time Transport Protocol )
    RTP
    A General view of the real-time protocols
    • Stream description : SDP, SMIL
    Describe the session and content
    • Stream Control : RTSP
    Remote control the session
    • Session Handling : SIP, SAP, SDP
    Session initiation, announcement, description
    • Media Transport : RTP and RSTP
    Send data and metadata
    RTP
    RTP
    Streaming : TCP or UDP ?
    Why not TCP?
    • Loss retransmission mechanism

    • Continuous rate fluctuation

    Why not UDP


    • Too simple and not reliable

    • Does not take account for take account for the lost packet and delayed
    packet
    RTP
    • UDP is used to carry pure audio streaming
     Like audio and VoIP

    • TCP is used for streaming; large buffer delay


     Ok for one way application, like watching clips from YouTube
     Not ok for interactive application, like video conference
    RTP
    • Real-time Transport protocol (RTP)
     A framing protocol for real-time applications
     Does not define any QoS mechanism for real-time delivery

    • Real-time control protocol (RTCP)


     It’s a companion control protocol
     Does not guarantee anything either
     Provides feedback

    The main purpose is to aid the application to know something about the
    network and accordingly adapt its behavior.
    RTP
    Design goal and philosophy
    • Flexible
     Provide mechanisms, do not dictate algorithm
     Instantiations for H261, MPEG 1/2/…
    • Protocol neutral
     UDP/IP, Private ATM networks…
    • Scalable
    • Unicast, Multicast, from 2 to …
    • Separate control/data
     Some functions may be taken over by conference control protocol
    (e.g. RTSP)
    RTP Requirements
    • Meet the requirements of
     Interactive multimedia applications (strict real-time constraints)
     Streaming application (not so strict)
    • Allow similar applications to talk to each other
     Negotiate on coding schemes
    • Help recipients determine timing relationships among data
     Time-stamping
     Synchronization of multiple-media
    RTP Requirements
    • Provide an indication of packet loss
     Application needs to handle it
    • Provide an indication of “frame boundary”
     Differs with respect to applications
     Audio—talkspurt
     Video– I/P/B frames
    • Provide a generic identity for senders
     Independent of IP address
    • All of this without to much header
    RTP Requirements
    • Application-level framing
     An application knows best what it needs
     Have different profiles (formats) for different applications
     Advantage : app-specific flexibility
    • End-to-end principle
     End systems to take responsibility of providing service irrespective
    of N/W capabilities
     Intelligence in the applications

    • Provide a generic identity for senders


     Independent of IP address
    • All of this without to much header
    RTP Features
    • RFC 3550
    • RTP runs in end systems
    • RTP packets are encapsulated in UDP segments
    • Manage delivery of real-time (RT) data
     Specifies packet structure for packets carrying audio, video data
    • Interoperability
     If two VoIP applications run RTP, they can run together
    RTP Features
    • RPT packet provides
     Payload type identification
     Source identifier
     Sequence number for loss detection
     Time-stamping for timing recovery
     Marker for significant events in the data stream (based on the
    application)
    RTP Functions
    Data Functions
    • Content labeling
     Source identification
     Loss detection
     Re-sequencing
    • Timing
     Intra-media Synchronization
     Removes jitter with playout buffers
     Inter-media Synchronization
     Lip synchronization between audio-video
    RTP
    RTP libraries provide transport-layer interface that extends UDP
    • Port number, IP address
    • Payload type identification
    • Packet sequence numbering
    • Time-stamping

    • No fixed UDP port : Negotiated out of band


    • UDP port for RCTP : UDP port for RTP+1
    • Usually one media per RTP session (i.e. port pair)
    How RTP Works

    RTP
    UDP RTP Video
    IP Header
    Header Payload
    Header

    UDP RTP
    RTP Audio
    IP Header Header
    Payload
    Header

    ●Video and audio payloads are sent separately


    ●Uses sequence number to synchronise audio and
    video once received
    RTP

    RTP operation
    RTP Header Format
    RTP Header Format
    Payload type (7 bits)
    • Indicates the type of encoding currently being used
    • If the sender changes encoding during call, sender informs receiver via
    payload type field
    Payload type 0 : PCM -law, 64Kbps
    Payload type 3 : GSM , 13Kbps
    Payload type 7 : LPC, 2.4Kbps
    Payload type 26 : Motion JPEG
    Payload type 31 : H.261
    Payload type 33 : MPEG2 video

    Sequence (16 bits)


    • Increment by one for each RTP packet sent
    • Detect packet loss,
    • Restore packet sequence
    RTP Header Format
    Time stamp
    • To enable different media stream to be synchronized

    • A counter of ticks

    • Granularity of ticks is app specific

    • Represents time at which first sample in pkt was generated

    • Difference in time-stamps used for synchronization


    RTP Header Format
    Synchronization source ID (SSRC)
    • Uniquely identifies single source of RTP stream

    Contributing source ID (CSRC)


    • When number of RTP streams pass through mixer
    RTP Header Format
    Time stamp field (32 bits long)
    • Sampling instant of first byte in this RTP data packet

    • For audio, time-stamp clock increments by one for each sampling


    period

    • If application generates chunks of 160 encoded samples (20ms), time


    stamp increases by 160 for each RTP packet when source is active
    (sequence no increased by 1)

    • Time-stamp clock continues to increase at constant rate when source


    is inactive (i.e. for talkspurt)
    RTP Header Format
    SSRC field (32 bits)
    • Identifies source of RTP stream.
     E.g. sending 64Kbps PCM-encoded voice over RTP
     Application collects encoded data in chunks
     E.g. every 20ms; 160 bytes in a chunks
     Audio chunks +RTP header from RTP packet, which is
    encapsulated in UDP segment
    • Each stream in RTP session has distinct SSRC
    • RTP header indicates type of audio encoding in each packet
     Sender can change encoding during conference
    • RTP header also contains sequence numbers, timestamps
    RTP multiplexing using SSRC
    • One RTP session between VoIP gateways
     Many phone calls between branch offices
     Multiplexing using different SSRC ID within the RTP
    session
    RTP Mixer
    • To combine the streams from different sources
    • Mixer becomes the SSRC
    • Others are CSRC
    RTP Packet Validation
    • There is no protocol field
    • Need to check for version field or some fields in stream of packets
    • Per-flow checking (i.e. look for constant SSRC and increasing)
    RTP & TCP/IP Model
    Internet Protocol Suite

    Application Layer

    BGP · DHCP · DNS · FTP · HTTP ·IMAP · IRC · LDAP ·


    MGCP · NNTP ·NTP · POP · RIP · RPC · RTP · SIP
    ·SMTP · SNMP · SSH · Telnet ·TLS/SSL · XMPP ·
    Transport Layer

    TCP · UDP · DCCP · SCTP · RSVP ·ECN

    Internet Layer

    IP (IPv4, IPv6) · ICMP · ICMPv6 · IGMP ·IPsec ·

    Link Layer

    ARP/InARP · NDP · OSPF ·Tunnels (L2TP) · PPP ·


    Media Access Control (Ethernet, DSL, ISDN, FDDI)
    RTP

    IPTV Protocol Stack


    RTP

    SIP - Protocol Stack


    RTP

    The functional sections of the IP stack


    Thanks

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