Chapter 1 outline

1
Chapter 1 outline

• 1.1 Multimedia Networking Applications
• 1.2 Audio and Video Basics, Formats,
  andStructures
• 1.3 Streaming Stored Audio and Video
• RTSP
• 1.4 Real-time Multimedia Internet Phone Case
  Study

• 1.5 Protocols for Real-Time Interactive
  Applications
• RTP,RTCP
• SIP
• 1.6 Beyond Best Effort
• 1.7 Scheduling and Policing Mechanisms
• 1.8 Integrated Services
• RSVP
• 1.9 Differentiated Services

2
Real-Time Protocol (RTP)

• RTP specifies a standardized packet structure for
  packets carrying audio and video data.
• Found in RFC 1889.
• RTP packet provides
• payload type identification
• packet sequence numbering
• time-stamping

• RTP runs in the end systems.
• RTP packets are encapsulated in UDP segments.
• A key goal of RTP is interoperability if two
  Internet phone applications run over RTP, then
  they may be able to work togetherwithout
  problems.

3
RTP Runs on Top of UDP

• RTP libraries provide a transport-layer interface
  that extends UDP
• port numbers, IP addresses
• payload type identification
• packet sequence numbering
• time-stamping

4
RTP Example

• Consider sending 64 kbps PCM-encoded voice over
  RTP.
• Application collects the encoded data in chunks.
  For example, every 20 msec or 160 bytes in a
  chunk.
• The audio chunk along with the RTP header
  (normally 12 bytes) form the RTP packet, which is
  encapsulated into a UDP segment.

• The RTP header indicates the type of audio
  encoding in each packet.
• The sender can change the encoding during a
  conference.
• RTP header also contains sequence numbers and
  timestamps.

5
RTP and QoS

• RTP does not provide any mechanism to ensure
  timely delivery of data or to 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.
• In fact, routers make no distinction between UDP
  packets carrying RTP payloads and those that are
  not. They are all treated the same!
• So, you can get packet loss, corruption, and
  ordering problems, just like UDP.

6
RTP Header

• Payload Type (7 bits) Indicates type of
  encoding currently being used. If the
  sender changes encoding in middle of conference,
  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 14, MPEG Audio
• Payload type 26, Motion JPEG
• Payload type 31. H.261
• Payload type 33, MPEG2 video
• 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. (For example, detecting a gap in
  sequence numbers can cause loss concealment
  algorithms to kick in.)

7
RTP Header (2)

• Timestamp field (32 bits long). Reflects the
  sampling instant of the first byte in the RTP
  data packet. Each timestamp is derived from a
  sampling clock at the sender.
• For audio, timestamp clock typically increments
  by one for each sampling period (for example,
  each 125 usecs for a 8 KHz sampling clock).
• If an application generates chunks of 160 encoded
  samples, then the timestamp increases by 160 for
  each RTP packet when the source is active. The
  timestamp clock continues to increase at a
  constant rate even when the source is inactive.
• Synchronization Source (SSRC) field (32 bits
  long). Identifies the source of the RTP stream.
  Each stream in a RTP session should have a
  distinct SSRC. This is not the IP address of the
  sender, but a number that the source assigns
  randomly when a new stream is started.

8
Writing RTP-Aware Applications

• Two Main Approaches
• The application developer writes the code to do
  the RTP encapsulation at the sender side and
  extraction at the receiver side.
• Use existing RTP libraries!
• Libraries and packages exist for most languages
  (C, C, Java, and so on)
• Why reinvent the wheel?

9
Real-Time Control Protocol (RTCP)

• Works in conjunction with RTP (its also defined
  in RFC 1889 with RTP).
• Each participant in an RTP session periodically
  transmits RTCP control packets to all other
  participants using IP multicasting.
• Each RTCP packet contains sender and/or receiver
  reports and is encapsulated in a UDP segment.
• These reports provide statistics and information
  that might be useful to the application.

• Statistics include number of packets sent, number
  of packets lost, interarrival jitter, etc.
• The RTCP standard does not dictate what should be
  done with this feedback it is up to the
  application developer to decide.
• For example, this feedback can be used to control
  performance.
• A sender in the application may modify its
  transmissions based on this feedback.

10
RTCP - Continued

• For an RTP session, there is typically a single
  multicast address all RTP and RTCP packets
  belonging to the session use the multicast
  address.
• RTP and RTCP packets are distinguished from each
  other through the use of distinct port numbers
  (the RTCP port number is set to be equal to the
  RTP port number, plus one).
• To limit traffic, each participant reduces its
  RTCP traffic as the number of conference
  participants increases.

11
RTCP Packets

• Source description packets
• E-mail address of sender, sender's name, SSRC of
  associated RTP stream.
• Provide mapping between the SSRC and the
  user/host name.

• Receiver report packets
• SSRC of the RTP stream, fraction of packets lost,
  last sequence number, average interarrival
  jitter.
• Sender report packets
• SSRC of the RTP stream, the current timestamp and
  wall clock time, the number of packets sent, and
  the number of bytes sent.

12
Synchronization of Streams Using RTCP

• RTCP can synchronize different media streams
  within a RTP session.
• Consider a video conferencing application for
  which each sender generates one RTP stream for
  video and one for audio.
• Timestamps in RTP packets are tied to the
  individual video and audio sampling clocks.
• They are not tied to the wall-clock time, or each
  other!

• Each RTCP sender-report packet contains (for the
  most recently generated packet in the associated
  RTP stream)
• The timestamp of the RTP packet.
• The wall-clock time for when the packet was
  created.
• Receivers can use this association to synchronize
  the playout of audio and video.

13
RTCP Bandwidth Scaling

• Problem
• What happens when there is one sender and many
  receivers? RTCP reports scale linearly with the
  number of participants and would match or exceed
  the amount of RTP data! More overhead than
  useful data!
• Solution
• RTCP attempts to limit its traffic to 5 of the
  session bandwidth to ensure it can scale!
• RTCP gives 75 of this rate to the receivers and
  the remaining 25 to the sender.

• Example
• Suppose one sender, sending video at a rate of 2
  Mbps. Then RTCP attempts to limit its traffic to
  100 Kbps.
• The 75 kbps is equally shared among receivers
• With R receivers, each receiver gets to send
  RTCP traffic at 75/R kbps.
• Sender gets to send RTCP traffic at 25 kbps.
• Participants determine their RTCP packet
  transmission period by calculating the average
  RTCP packet size (across the entire session) and
  dividing by their allocated rate.

14
Session Initiation Protocol (SIP)

• Comes from the IETF (RFC 3261).
• 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 would like to be able to reach the callee, no
  matter where the callee roams, no matter what IP
  device the callee is currently using.
• Doing this in practice is quite difficult!

15
SIP Services

• Setting up a call
• Provides mechanisms for a caller to let the
  callee know that the caller wants to establish a
  call.
• Provides mechanisms so that caller and callee can
  agree upon the media type and encoding to be used
  in the call.
• Provides mechanisms to end calls when done.

• Determines the current IP address of callee.
• Maps mnemonic identifier (name, e-mail address,
  etc.) to the current IP address.
• Call management
• Add new media streams during call.
• Change encoding during call.
• Invite others.
• Transfer and hold calls.

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Setting Up a Call to a Known IP Address

• Alices SIP invite message indicates her port
  number IP address. Indicates encoding that
  Alice prefers to receive (PCM µlaw).
• Bobs 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 5060.

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Setting Up a Call (More)

• Codec negotiation
• Suppose Bob doesnt 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.

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Example of SIP Message

• INVITE sipbob_at_domain.com SIP/2.0
• Via SIP/2.0/UDP 167.180.112.24
• From sipalice_at_hereway.com
• To sipbob_at_domain.com
• Call-ID a2e3a_at_pigeon.hereway.com
• Content-Type application/sdp
• Content-Length 885
• cIN IP4 167.180.112.24
• maudio 38060 RTP/AVP 0
• Notes
• HTTP/SMTP like message syntax.
• sdp session description protocol
• Call-ID is unique for every call.

• Here we dont know
• Bobs IP address.
• Intermediate SIPservers will be necessary.

• Alice sends and receives SIP messages using
  the SIP default port number 5060.
• Alice specifies in Viaheader that SIP client
  sends and receives SIP messages over UDP.

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Name Translation and User Location

• Caller wants to call callee, but only has
  callees name or e-mail address.
• Need to get IP address of callees current host
• User can be mobile.
• DHCP protocol may be dynamically assigning
  different addresses for each session.
• User can have multiple IP devices (PC at home or
  work, PDA, etc.).

• Result can be based on
• Time of day (work, home).
• Caller (dont 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

20
SIP Registrar

• When Bob starts a SIP client, the client sends a
  SIP REGISTER message to Bobs registrar server
  (similar function needed by Instant Messaging).
• The SIP registrar is very much like a DNS
  authoritative name server (translates fixed human
  identifiers to potentially dynamic IP addresses).

Register Message

• REGISTER sipdomain.com SIP/2.0
• Via SIP/2.0/UDP 193.64.210.89
• From sipbob_at_domain.com
• To sipbob_at_domain.com
• Expires 3600

21
SIP Proxy

• Alice sends INVITE message to her proxy server.
• This message contains the address
  sipbob_at_domain.com.
• This proxy is somehow responsible for routing SIP
  messages to the callee.
• Possibly through multiple proxies.
• The callee sends its response back through the
  same set of proxies.
• Proxy returns SIP response message to Alice
• This response contains Bobs IP address.
• Note proxy is analogous to local DNS server.

22
SIP Example
Caller jim_at_umass.edu with places a call to
keith_at_upenn.edu (1) Jim sends INVITEmessage to
umass SIPproxy. (2) Proxy forwardsrequest to
upenn registrar server. (3) upenn server
returnsredirect response,indicating that it
should try keith_at_eurecom.fr
(4) umass proxy sends INVITE to eurecom
registrar. (5) eurecom registrar forwards INVITE
to 197.87.54.21, which is running keiths SIP
client. (6-8) SIP response sent back (9) media
sent directly between clients. Note There is
also a SIP ACK message, which is not shown.
23
Comparison with H.323

• H.323 is another signaling protocol for
  real-time, interactive multimedia.
• 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. It works with RTP, but
  does not mandate it. It can be combined with
  other protocols and services.

• H.323 comes from the ITU (telephony).
• SIP comes from IETF, so it borrows many of its
  concepts from HTTP. SIP has a Web flavor,
  whereas H.323 has a telephony flavor.
• H.323 is large and complex (because it is
  complete), whereas SIP uses the KISS principle
  Keep it simple stupid.

24
Chapter 1 outline

• 1.1 Multimedia Networking Applications
• 1.2 Audio and Video Basics, Formats,
  andStructures
• 1.3 Streaming Stored Audio and Video
• RTSP
• 1.4 Real-time Multimedia Internet Phone Case
  Study

• 1.5 Protocols for Real-Time Interactive
  Applications
• RTP,RTCP
• SIP
• 1.6 Beyond Best Effort
• 1.7 Scheduling and Policing Mechanisms
• 1.8 Integrated Services
• RSVP
• 1.9 Differentiated Services

25
Improving QoS in IP Networks

• Thus far making the best of best effort
• Future next generation Internet with QoS
  guarantees
• Integrated Services firm guarantees
• RSVP signaling for resource reservations
• Differentiated Services differential guarantees
• simple model for sharing and congestion
  studies

26
Principles for QoS Guarantees

• Example 1Mbps IP phone and FTP share 1.5 Mbps
  link.
• Bursts of FTP can congest the router and cause
  audio loss.
• We want to give priority to audio over FTP.

Principle 1
Packet marking is needed for the router to
distinguish between different classes, and a new
router policy is needed to treat packets
accordingly.
27
Principles for QoS Guarantees (More)

• What if applications misbehave (for example,
  audio sends its data higher than its declared
  rate).
• Policing force source adherence to bandwidth
  allocations.
• Marking and policing at network edge
• Similar to ATM UNI (User Network Interface).

Principle 2
Provide protection (isolation) for one flow from
others.
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Principles for QoS Guarantees (More)

• Allocating fixed (non-sharable) bandwidth to
  flow inefficient use of bandwidth if a flow
  doesnt use its allocation.

Principle 3
While providing isolation, it is desirable to use
resources as efficiently as possible.
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Principles for QoS Guarantees (More)

• Basic fact of life a network cannot support
  traffic demands beyond its link capacity.

Principle 4
Call Admission flow declares its needs, network
may block call (e.g., busy signal) if it cannot
meet needs.
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Summary of QoS Principles
Next, we look at mechanisms for achieving this .