IP Multicast

1
IP Multicast

• A convention to identify a multicast address
• Each node must translate between an IP multicast
  address and a list of networks that contain
  members of this group
• A router must translate between an IP multicast
  address and a subnetwork multicast address to
  deliver a multicast IP datagram on the
  destination network
• A mechanism for an individual host to inform
  routers (on the same network as itself ) of its
  inclusion or exclusion from a multicast group

2
IP Multicast (Contd.)

• Routers need to know which subnetworks include
  members of a given multicast group.
• Routers need sufficient information to calculate
  the shortest path to each network containing
  group members.
• A routing algorithm is needed to calculate
  shortest paths to all group members.
• Each router must determine multicast routing
  paths on the basis of both source and destination
  addresses.

3
IP Multicast (Contd.)

• IGMP (Internet Group Management Protocol) or
  ICMPv6 (Internet Control Message Protocol) is
  used by hosts and routers to exchange multicast
  group membership information
• MOSPF (Multicast Extension to Open Shortest Path
  First) enables routing of IP multicast datagrams
• Each router attached to a LAN uses IGMP to
  maintain a current picture of local group
  membership. Periodically each router floods group
  membership information to all other routers in
  its area.
• Using Dijkstras algorithm, each router
  constructs the shortest-path spanning tree from
  source network to all networks containing members
  of multicast group (done only on demand).
• Unique spanning tree for a given source node
• Interarea multicasting

4
Streaming on the Web

• The contents of a compressed audio and/or video
  file are played out as they are being received. A
  playout buffer helps to smooth the variations in
  the time between each received packet in the
  stream (delay jitter).
• Media player acts as the interface between the
  incoming compressed media bitstream and the
  related sound and/or video output cards.
• For video, the browser first creates a window in
  the web page and passes the coordinates of the
  window to the selected video media player.
• Video media player initializes the video card
  with the assigned coordinates and decompresses
  the video bitstream from the playout buffer and
  passes it to the video card for rendering
• Media player consists of two parts playout
  functions and control functions for interactivity

5
Sequence

• Browser sends a HTTP GET request to the web
  server
• Web server responds by returning the contents of
  the meta file to the browser
• Browser determines the media player to invoke
  from the content type field and passes the
  contents of the meta file to the selected media
  player
• Control part of the media player requests control
  part of the streaming server to start a new
  session by sending RTSP SETUP request message
  (including the port number allocated to RTP). In
  response the control part of the server returns
  an RTSP accept message (with a unique session
  identifier)
• Control part of the media player sends a RTSP
  PLAY request message the control part on the
  server side initiates the access and transmission
  of the packet stream using the allocated port
  number of RTP in the header of each UDP datagram
• RTSP TEARDOWN request message is used when the
  user activates the quit/end button

6
RTP

• The timing information required by the receiver
  to output the received packet stream at the
  required rate is provided by RTP
• RTCP is used to synchronize the two media streams
  prior to carrying out the decoding the decoding
  operation
• RTP functions (1) detecting missing packets and
  compensating for lost packets as well as delay
  variations (2) reconstructing bitstream
  (reordering the packets)

7
RTP Packet Format

• CSRC -- contributing source used in a multicast
  call/session each source is uniquely identified
  by means of a 32-bit identifier (which is
  typically the IP address of the source)
• During a multicast session, the packet stream
  from multiple sources may be multiplexed together
  for transmission purposes by a device known as a
  Mixer.
• Resulting RTP packet may contain blocks/frames of
  digitized information from multiple sources
• To enable the receiver to relate each block/frame
  to the appropriate participant, the CSRC
  identifier for each block/frame is included in
  the header of the new packet
• CC field (4 bits) indicates the number of CSRC
  identifiers present in the packet

8
RTP Packet Format (Contd.)

• Payload type field indicates the type of encoder
  that has been used to encode the data in the
  packet. Since each packet contains this field,
  the encoder type used can be changed from packet
  to packet during a call depending on n/w load.
• Each packet contains a sequence number which is
  used to detect lost or out-of-sequence packets.
• In the case of a lost packet, the contents of the
  last correctly received packet are used in its
  place.
• The effect of out of sequence packets is overcome
  by buffering a number of packets before playout
  of the data they contain starts.

9
RTP Packet Format (Contd.)

• The value of the time stamp field indicates the
  time reference when the packet was created.
• It is used to determine the current mean
  transmission delay time and the level of jitter
  that is being experienced.
• Along with the delay information, the number of
  lost packets forms part of the current QoS of the
  path through the n/w.
• RTCP periodically sends this info to the sending
  RTP source so that the sending RTP may modify the
  resolution of the compression algorithm if the
  n/w load changes. Level of jitter is used to
  determine the size of the playout buffer.
• Synchronization source (SSRC) identifier
  identifies the source device that produced the
  packet contents. Receiving RTP uses the SSRC to
  relay the reconstructed bitstream to the related
  output device interface.

10
RTCP

• RTCP operates along side of RTP and shares
  information with it
• Each RTCP has a different (UDP) port number
  associated with it so that it can operate
  independently of RTP
• Periodic messages (RTCP packet) exchanged with
  the RTP hosts/clients
• common system time clock for media
  synchronization
• QoS reports computed by the receiving RTPs
• participation reports
• participation details

11
RTCP Messages

• RTCP from all call participants periodically
  exchange messages with one another. Each message
  is sent in a RTCP packet to the same network
  address as the RTP to which the message relates.
• For applications that involve separate audio and
  video streams, a common system time clock is used
  for synchronization purposes.
• The number of lost packets, the level of jitter,
  and the mean transmission delay are continuously
  computed by each RTP for the packet streams they
  receive from all other contributing sources. RTCP
  passes this information to all other RTCPs.
• Participation reports used when a participant
  leaves the call.
• Participation details includes information such
  as the name, email address, phone number and so
  on of each participant.

12
Internet Telephony

• When a PC connected to the Internet needs to make
  a call to a telephone that is connected to a
  PSTN/ISDN, an internetworking unit known as
  telephony gateway is used.
• PC user sends a request to make a telephone call
  to the preallocated telephony gateway using the
  g/ws Internet address.
• The g/w requests the source PC for the phone
  number of the called party.
• Then the source g/w initiates a session w/ the
  telephone g/w nearest to the called party using
  the IP address of the g/w.
• Called g/w initiates a call to the recipient
  telephone using its phone number and standard
  call setup procedure of the PSTN/ISDN.
• Assuming the called party answers, the called g/w
  then signals back to the PC user -- through
  source g/w -- that the call can commence.
• PC to PC h/w and s/w to convert speech signal
  from the microphone to packets on input and back
  again prior to output to the speakers.

13
Session Initiation Protocol (SIP)

• Provides services for user location, call/session
  establishment, and call participation management
• A simple request-response protocol -- both the
  request and response are made through an
  application program called the user agent (UA)
  which maps the request and its response into the
  standard message format used by SIP
• Each UA comprises two parts, a UA Client (UAC),
  which enables the user to send request messages
  (to initiate the setting up of a call/session)
  and a UA server (UAS) which generates the
  response message

14
Call/Session Set Up

• SIP name/address is similar to an email
  name/address. A single user may have a number of
  alternative locations/addresses
• Calling host sends an INVITE request message to
  the local proxy server (PS -A) the proxy server
  (PS-A) obtains the IP address of the proxy server
  (PS-B) for the called host and the SIP in the
  proxy server (PS-A) sends the INVITE request
  message to PS-B. PS -B determines that the user
  is currently logged in (gets name/address from
  the SIP message) and the IP address of the
  called host. If the called host can accept the
  call, an INVITE response message is returned over
  the same path. Receiving this the SIP in the
  calling host returns an ACK and the two
  users/hosts start exchanging the info.

15
Session Description Protocol (SDP)

• To describe the different media streams involved
  in a call/session
• Described in each SIP message body in text
  format
• media streams (list of media types and format in
  each SIP INVITE request message and a modified
  version of that in the INVITE response message)
• stream addresses (destination address and UDP
  port number for sending and/or receiving each
  stream)
• start and stop times (for broadcast sessions)