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Multimedia Networking
Network Requirements and Protocols
Ahrar Naqvi
[email protected]
Agenda
• Overview of media characteristics from a
networking standpoint
• Network requirements for multimedia
communications
• Network architectures, techniques and
protocols
Multimedia Classification
• Real Time: Require bounds on end-to-end
packet delay & jitter. Subdivided into:
– Discrete Media: MSN/Yahoo Messenger, Stock
quotes
– Continuous Media: Continuous message stream with
inter-message dependency. Further divided into:
• Delay Tolerant e.g Internet webcast
• Delay Intolerant e.g. audio, video streams in conferencing
systems
• Non-Real Time: No strict delay constraints (e.g.
text, image files)
– May be highly sensitive to errors
Key Media Characteristics from
Networking Standpoint
• Bandwidth usage
• Sensitivity to error
• Real-time nature
Text
• Bandwidth requirements depend on size
– Can be easily reduced by compression
techniques
• Some text applications require complete
freedom from loss & errors – use TCP.
E.g. FTP
• Others are error and loss tolerant – use
UDP e.g. instant messaging
Audio
• Bandwidth requirements depend on dynamic range
and/or spectrum
– Narrowband speech (300-3300Hz)
• 6.4 Kbps (G.723.3) to 64 Kbps (G.711)
– Wideband audio (CD quality music) 10-220 KHz
• 112-128 Kbps (MP3)
• Can tolerate 1-2% packet loss
• Real-time nature depends on extent of interactivity
– VoIP requires strong bounds on delay/jitter (Real-Time
Intolerant)
• < 250 ms end to end delay
– Internet Webcast is more delay/jitter tolerant (Real-Time
Tolerant)
Graphics and Animation
• Examples: Digital images, flash presentations
• Large in size but lend themselves well to
compression
• Progressive compression techniques enable
image to be initially displayed in low-quality and
gradually improved as more information is
received
• Error-tolerant and can sustain packet loss
provided application knows how to deal with
packet loss
• No real-time constraints
Video
• High bandwidth requirements
• Efficient compression schemes
– MPEG-I (1.2 Mbps) VCR quality compression
– MPEG-II (3-100 Mbps) broadcast quality video, HDTV
– MPEG-IV (64 Kbps) for low bandwidth video
compression; supports audio, video, graphics,
animation, text
– H.261 (px64 Kbps) video over ISDN
– H.263 (18-64 Kbps) video over POTS
• Error requirements and real-time characteristics
similar to audio
Network Requirements
• Traffic Requirements: Have implications for basic
Internet infrastructure
– Limits on Delay & Jitter
– Bandwidth
– Reliability
• Functional Requirements: Require enhancements to
TCP/IP stack in the form of additional network protocols
– Multicasting
• Single source of communication with multiple simultaneous
receivers
• Can be one way (internet radio) or two way (multi-party audio/video
conferencing)
– Security
– Mobility
– Session Management
Delay Related Metrics
• Maximum end to end delay
• Delay variance
– Jitter: non-monotonic variation in delay in given
stream
• For a video stream jitter would result in a shaky picture
• Jitter can be removed by buffering at the receiver side
– Skew: constantly increasing difference between the
expected arrival time and the acutal arrival time
• For a video stream skew could be a slower or faster moving
picture
Delay: Packet Processing Delay
• Constant amount of delay at both source and destination
– A/D, D/A conversion time and time taken to packetize it through
different layers of protocols
• Typically a characteristic of the operating system and the
multimedia application
• Delay can become significant under high load conditions
• Reductions in delay imply software enhancements
including use of multimedia operating systems that
provide enhanced resource, file and memory
management with real-time scheduling
Delay: Packet Transmission Delay
• Time taken by the physical layer at the source to
transmit packets. Depends on
– Number of active sessions. Typically physical layer
processes packets in FIFO order. Delay can become
significant if OS does not support real-time scheduling
for multimedia traffic
– MAC access delay: Widespread Ethernet networks
cannot provide any firm guarantees on medium
access delay due to inherent indeterminism in
CSMA/CD (carrier sense multiple access/collision
detection). Isochronous Ethernet (802.9) integrated
voice data LAN and demand priority Ethernet
(802.12) provide QoS but market potential remains
low
Delay: Propagation Delay
• Flight time of packets – limited by speed of
light. Can’t do anything about it
• For a distance of 20,000 km this would be
about 0.067 sec
• Significant part of a desirable ~200 msec
delay budget
Delay: Routing and Queuing Delay
• Best-effort Internet treats every packet equally
• Packets arriving at a queue have to wait a
random amount of time depending on current
router load
• Delay is variable and is the major contributor to
jitter
• Techniques to reduce this include
– IntServ
– MPLS
– DiffServ
Bandwidth Requirements
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Multimedia traffic streams have high bandwidth
Uncontrolled transmissions at high rates can cause heavy congestion in the
network
Elastic applications that use TCP take advantage of built in congestion
control
Most multimedia applications use UDP for transmitting media streams
It is left to the discretion of the application to dynamically adapt to
congestion
To remove these shortcomings an enhanced internet service model would
require
– Admission control: application must first get permission from some authority to
send traffic at a given rate with given traffic characteristics
– Bandwidth reservation: if admission is given, appropriate resources (buffers,
bandwidth) will get reserved along the path
– Traffic policing mechanisms: to ensure that applications do not send data at a
rate higher than what was negotiated
Bandwidth Related Metrics
• Cost of protocol processing operations relates
more directly to packet processing rate than to
the bandwidth in terms of bit rate
• In addition to a bit-rate oriented specification of
bandwidth, packet processing rate is an
important metric
• Cost of packet processing is largely dependent
on number of packets and less so on packet size
• Packetization related metrics include maximum
and average packet size and packet rate
Reliability
• Pertains to loss and corruption of data
• Can be measured in terms of loss
probability
• Requires methods for dealing with
erroneous/lost data
Error Correction
• Sender Based Repair
• Active Repair - Automatic retransmission request (ARQ)
– Suitable for error intolerant applications
• Passive Repair
– Interleaving
– FEC: Forward Error Correction.
» Media Independent – independent of the content/nature of
the stream
» Media Dependent - use knowledge of the stream in the
repair process
• Error Concealment (Receiver Based Repair)
FEC
• Introduce repair data in traffic from which lost packets may be
recovered
• Media Independent: use block or algebraic codes to produce
additional packets which aid in loss recovery
– Each code takes a codeword of k data packets and generates n-k
additional check packets
– i-th bit in check packet is generated from the i-th bits of each associated
data packet
• Parity Coding: XOR is applied across groups of packets to generate parity
packets
• Reed-Solomon Coding: Based on properties of polynomials over particular
number bases
– Take a set of codewords and use these as coefficients of a polynomial f(x)
– The transmitted codeword is determined by evaluating the polynomial for all
nonzero values of x over the number base
– Disadvantage: Cause additional delay, increase bandwidth usage and
exacerbate congestion
Media Dependent FEC
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Exploit media characteristics
For audio, could send each unit of audio in
multiple packets
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Primary encoding: first transmission
Secondary encoding: additional transmissions
Secondary encoding could be of lower bandwidth
and quality than the primary coding
May not be necessary to transmit FEC for every
packet due to nature of media
Advantage: low latency – only single packet
delay added
–
Suitable for interactive applications
* A Survey of Packet Loss Recovery Techniques for
Streaming Audio, Colin Perkins et al IEEE
Network Sep/Oct 1998
Interleaving
• Can be used when media unit size is smaller than packet size (as
may be the case with audio) and end-to-end delay is not important
• Units are resequenced before transmission so that originally
adjacent units are separated by a guaranteed distance and returned
to original order at the receiver
• Disperses the effect of packet loss – loss of a single packet would
causes multiple smaller gaps among original media units
• In case of audio a phoneme originally encapsulated in one packet
would get split across multiple packets
• Loss of small parts of several phonemes is easier to deal with than
loss of entire phonemes
• Disadvantage: increased latency – not well suited for interactive
applications
• Advantage: does not increase bandwidth usage – does well for noninteractive use
Error Concealment
• Producing a replacement for a lost packet which is
similar to the original
• Work for relatively small loss rates (< 15%) and for small
packets (4-40 ms)
• Types in increasing order of computational cost and
improved performance:
– Insertion based: insert a fill-in packet that contains silence, noise
or a repitition of an adjacent packet
– Interpolation-based: some form of pattern matching and
interpolation to derive the missing packet (waveform, pitch or
timescale based)
– Regeneration-based: derive decoder state from packets
surrounding the loss and generate a lost packet from that (model
based recovery)
Error Recovery for Different
Applications
• Non-interactive Applications
– Multicasts (e.g. radio)
• Interleaving is suitable (bandwidth efficient, though
high latency)
• Use error concealment – repetition with fading
• Media-independent FEC better than a
retransmission based scheme
• Interactive Applications (e.g. IP telephony)
– Media Dependent FEC
– Error concealment using packet repetition
Admission Control
• Pro-active form of congestion control
• Takes requested traffic description as input
including (in terms of leaky bucket
parameters (b,r))
– Maximum burst size ( b = bucket size)
– Peak rate
– Average rate
– Decides to accept or reject a flow including
consideration of impact to existing flows
Admission Control
• Leaky bucket parameters provide a very loose
upper bound on the traffic rate
• When traffic becomes bursty network utilization
can become very low if admission control is
solely based on parameters provided at call
setup time
• Admission control unit must also use
measurements of current network load and
packet delay in its admission decisions
Traffice Shaping/Policing
• Token bucket algorithm is used for traffic
shaping. Limits the average rate and allows a
degree of burstiness.
– Token bucket depth ‘b’ in which tokens are collected
at rate ‘r’
– When bucket becomes full extra tokens are dropped
– Source can send data only if it can grab and destroy
sufficient tokens from the bucket
• Leaky bucket algorithm is used for traffic
policing, in which excessive traffic is dropped
– Bucket depth ‘b’ with hole at the bottom
– If bucket is full extra packets are dropped
Packet Classification
• In order to prevent all packets from being
treated equally some mechanism to
distinguish between real-time and non-real
time packets is needed
• Done by packet marking e.g. use Type of
Service (ToS) field in IP header
• MPLS uses short labels
Packet Scheduling
• FIFO scheduling traditionally used in routers
needs to be replaced with more sophisticated
queuing
• E.g. priority queuing. Lower priority queues
served after higher priority queues
• Disadvantage: possible starvation of low priority
flows
• Weighted Fair Queuing has different queues for
different classes. However every queue is
assigned a certain weight. Packets in that queue
get a fraction of the total bandwidth proportional
to their weight
Packet Dropping
• Routers can randomly drop packets under
congestion
• This can be a problem since certain
packets may carry more information than
others
QoS Based Routing
• OSPF, RIP, BGP are best-effort routing protocols using
single-objective optimization algorithms: hop count or
line cost
• All traffic routed to shortest path can lead to congestion
on some links and leave other links under-utilized
• If link congestion is used to derive line cost such that
congested routes are costlier, such algorithms can cause
oscillations in the network, thus increasing jitter
• In QoS based routing paths are determined based on
some knowledge of resource availability in the network
as well as the QoS requirement of the flows
Integrated Services
• IntServ Internet Services model developed by IETF
• Requires applications to know their QoS requirements
beforehand and signal intermediate network to reserve
resources (bandwidth, buffers)
• Requires use of packet classifiers as well as packet
schedulers
• Almost exclusively concerned with controlling the
queuing component of end to end delay
• Has three service classes
– Guaranteed Service (RTI)
– Controlled Load (RTT)
– Best-effort
IntServ
• Flow descriptor specifies QoS requirements
– Filter spec – specifies information for the identifying a
packet with a given flow
– Flow spec – specifies traffic spec in terms of token
bucket parameters (Tspec) and QoS parameters
(Rspec) in terms of bandwidth, delay, jitter and packet
loss
• Resource Reservation Protocol (RSVP) used to
signal network nodes about required resources
RSVP
• The sender sends a PATH message to the
receiver specifying traffic characteristics
• Every intermediate router along the path
forwards the path to the next hop
determined by the routing protocol
• The receiver responds with RESV
message
Disadvantages of IntServ
• Routers having to maintain per-flow state
for every flow is a large overhead
• Does not scale in the core network
• Router state has to be refreshed at regular
intervals increasing traffic overhead
• However, RSVP has a place at the edge of
the network
DiffServ
• Removes some of the shortcomings of the
IntServ architecture
• Divides network into regions called DS domains.
Each domain is controlled by a single entity
• To provide service guarantees the entire path
between source and destination must be in
some DS domain
• Nodes in a DS domain can be of following types:
– Boundary node
– Interior node
DiffServ Boundary Node
• Performs admission control to limit the
number of flows in a domain
• Performs packet classification by marking
each packet with a service class called
“Behavior Aggregate”
• Each Behavior Aggregate is assigned an
8-bit code word called a DS code point
• IP ToS field is updated with the code-point
DiffServ Interior Node
• Lies completely within a DS domain. Connects
with other interior nodes or boundary nodes
within the domain
• Only performs packet forwarding
• Packets are forwarded according to some predefined rule associated with the packet class (as
indicated by the code point)
• These pre-defined rules are called Per-Hop
Behaviors (PHBs)
DiffServ PHB
• Two commonly used PHBs are
– Assured Forwarding (AF): Divides incoming traffic into
four classes where each class guarantees a minimum
bandwidth and buffer space
• Within each class packets are further assigned one of three
drop priorities
– Expedited Forwarding (EF): Departure rate of a traffic
class must equal or exceed the configured rate
• Queuing delay is guaranteed to be bounded
• Used to provide Premium Service
• Requires strict admission control and traffic policing
MPLS
• A router determines the next hop of a packet by doing a longest
prefix match of an IP destination address against entries in a routing
table
• This introduces some latency as routing tables can be very large
• Same process repeated for every packet even if these are in the
same flow
• IP switching gets around this:
– Short label is attached to every packet and is updated at every hop
– This label is used at the next hop as an index into the routing table to
get the next hop (happens in constant time) and next label
– Label is replaced and packet is forwarded to the next hop
– Lends itself to being done in (low-cost) hardware resulting in very high
speeds
MPLS
• Like DiffServ MPLS network is divided into domains with boundary
nodes called Label Edge Routers (LER) and interior nodes called
Label Switching Routers (LSR)
• Packets entering an MPLS domain are assigned a label at the
ingress LER and are switched inside a domain by a simple label
lookup
• Labels determine QoS
• Labels may get stripped off at egress LER and then get routed
conventionally outside the domain
• A sequence of LSR to be followed by a packet in an MPLS domain
is called a Label Switched Path (LSP)
• To guarantee QoS both source and destination have to be attached
to the same domain or the different domains have to have service
level agreements among them
MPLS
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A group of packets that are forwarded in the same manner are said to
belong to the same Forward Equivalence Class (FEC)- FECs form the basis
of service differentiation
No limit on number and granularity of FECs
Labels only have local significance
– Two LSRs agree upon using a particular label for a given FEC
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Necessary to do label assignments including label allocation and label to
FEC binding on every hop of the LSP before the traffic flow can use the LSP
Labels can be stacked in FILO order – can be used in tunneling applications
In a domain only the topmost label is used to make forwarding decisions
Can be useful in providing mobility
– Home agent can push a label on incoming packets and forward them to a foreign
agent
– Foreign agent pops of its label and forwards the packet to the destination mobile
host
MPLS Label Assignment
• Label assignment can be done in the following
ways:
– Topology-driven: LSPs for every possible FEC are
automatically set up between every pair of LSR (zero
call setup delay)
– Request driven: LSPs set up based on explicit
requests.
• RSVP can be used
– Traffic driven: LSPs set up only when LSR identifies
traffic patterns requiring a new LSP
• Traffic patterns not requiring an established LSP use the
normal routing method
• Combines advantages of above two
Multicasting – IP Multicast
• Can be done in several ways
• Send packets to multicast IP address (Class D)
• Hosts willing to receive multicast messages for particular multicast
groups inform immediate-neighboring routers using IGMP
• Multicast routers exchange group information using a variety of
algorithms:
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Flooding
Spanning tree
Reverse path broadcasting
Reverse path multicasting
• Protocols that use some of these algorithms include
– Distance Vector Muticast Routing Protocol (DVMRP)
– Multicast extension to Open Shortest Path First (MOSPF)
– Protocol Independent Multicast (PIM)
Multicasting – Overlay Network
(MBONE)
• Virtual network implemented on top of
some portions of the Internet
• Islands of multicast capable networks are
connected by virtual links called tunnels
– Tunneled packets are encapsulated as IP
over IP such that they look like normal unicast
packets to intervening routers
Application Layer Multicasting
• SIP and H.323 support multicasting
through a multi-point control unit that
provides mixing and conferencing
functionality
Session Management - Media
Description
• Enables application to distribute session
information
– Media Type
– Encoding Scheme
– Session Start Time
– Session Stop Time
– IP Addresses of involved hosts
Session Description Protocol
• SDP developed by IETF can be used to describe
media type, media encoding used for session
• More of a description syntax than a protocol –
augmented by SIP for media negotiation
• Media descriptions encoded in text format
• SDP message contains a series of lines called
fields with single letter abbreviations. Each field
has a <tag>=<value> format
Session Management - Session
Announcement
• Allows participants to announce future
sessions
• E.g. for Internet radio stations to distribute
information about scheduled shows
Session Announcement Protocol
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Used for advertising multicast conferences and sessions
SAP announcer periodically multicasts announcement packets to a wellknown multicast address and port (9875) with the same scope as the
session being announced
Recipients of announcement are also potential recipients of sessions being
advertised
Multiple announcers may announce a single session for more robustness
Announcement interval chosen to ensure total bandwidth used by
announcements is below a pre-configured limit
Each announcer is expected to listen to other announcements in order to
determine the total number of sessions being announced on a group
Involves large startup delay before complete set of announcements is heard
by a listener
Contains mechanisms for ensuring integrity, authenticating the origin and
encryption of announcements
Session Management - Session
Control
• Information in multiple media streams may be
inter-related
• Network must guarantee to maintain such
relationships – Multimedia Synchronization
• Can be achieved by putting timestamps in every
media packet
• Internet multimedia users may want to control
playback of continuous media – similar to what a
VCR or CD player provides
Session Control - RTP
• RTP runs on top of UDP
• Carries chunks of real-time (audio/video) data
• Provides
– Sequencing: sequence number in RTP header helps detect lost packets
– Payload Identification: payload identifier included in each RTP packet
describes encoding of the media
– Frame Indication: video and audio sent in logical units called frames. A
frame marker bit indicates the beginning and end of a frame
– Source Identification: To identify the originator of a frame in a multicast
session a Synchronization Source (SSRC) identifier
– Intramedia Synchronization: To compensate for different delay and jitter
for packets within the same stream RTP provides timestamps, which
are needed by play-out buffers
• Additional media information can be inserted using profile headers
and extensions
Real-Time Control Protocol - RTCP
• RTCP is a control protocol that works in
conjunction with RTP
• Provides useful statistics: packets sent, lost,
jitter, round-trip time
• Sources can use this to adjust their data rate
• Other information includes email address, phone
number, name – allow users to know the
identities of other users in the session
Real-Time Streaming Protocol
• RTSP is an out-of-band control protocol
that allows the media player to control the
transmission of the media stream including
functions such as
– Pause
– Resume
– Repositioning
– Playback
H.323
• Umbrella recommendation that specifies components, protocols and
procedures multimedia conferencing over a packet network
• Defines four components
– Terminals: These are the endpoints
– Gateway: For interoperation between clients using different H.32x
flavors
– Gatekeeper: Control functions including admission control, bandwidth
management, call routing
– Multi-point Control Unit: For point to multipoint conferencing capability
• Uses H.245 to determine common capabilities of terminals
• Two kinds of call control models
– Gatekeeper routed (preferred mode in carrier environments)
– Direct (not scalable)
H.323 Protocol Stack
H.323 Stack
• RTP & RTCP used for media
• H.225 RAS (Registration, Admission and Status) used by
endpoints and gateways to
– Gatekeeper discovery and registration
– Requesting call admission, bandwidth allocation
– Clearing a call
•
Q.931 signaling protocol is used for call setup and
teardown between two endpoints – lightweight version of
the ISDN protocol
• H.245 media control protocol is used for negotiating
media processing capabilities such as A/V codec
• T.120 is used for data-conferencing capabilities such as
whiteboard sharing, instant messaging
Session Initiation Protocol - SIP
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Application-layer signaling protocol for initiating, modifying and
terminating interactive sessions. Defined in RFC 3261
Does not define what a “session” is. Session is carried opaquely
in SIP messages.
Text-encoded protocol based on elements from HTTP and SMTP
“SIP supports five facets of establishing and terminating
multimedia communications:
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User location: determination of the end system to be used for
communication;
User availability: determination of the willingness of the called party to
engage in communications;
User capabilities: determination of the media and media parameters
to be used;
Session setup: "ringing", establishment of session parameters at both
called and calling party;
Session management: including transfer and termination of sessions,
modifying session parameters, and invoking services” (RFC 3261)
SIP – Key Capabilities
• A stateful SIP server can split or "fork" an
incoming call so that several extensions can be
rung at once
• The first extension to answer can take the call
• SIP can return different media types within a
single session
• Participants can be invited to existing sessions
• Media can be added (removed from) an existing
session
• Supports mobility
Elements of a SIP Network
• Three main elements in a SIP network
– User Agent: end device in a SIP network. User Agent Client
(UAC) initiates requests. User Agent Server (UAS) responds to
requests. Roles may change in the course of a session.
– Server: There are three main types
• Proxy: Receives requests from UAs or other proxy and forward the
request to another location
• Redirect: Receives a request from a UA or proxy and returns a
redirect response (3XX) indicating where the request should be
retried
• Registrar: Receives SIP registration requests and updates UA’s
information to a location server (e.g. LDAP server) or other
database
– Location Server: General term for a database. Non-SIP protocol
is used to interact with it.
SIP Methods
SIP Methods are commands supported by SIP:
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INVITE: Invites a user to a call
ACK: Used to facilitate reliable message exchange for INVITEs
BYE: Terminates a connection between users or declines a call
CANCEL: Terminates a request, or search, for a user
OPTIONS: Solicits information about a server's capabilities
REGISTER: Registers a user's current location
INFO: Used for mid-session signallingSIP responses
The following are SIP responses:
• 1xx Informational (e.g. 100 Trying, 180 Ringing)
• 2xx Successful (e.g. 200 OK, 202 Accepted)
• 3xx Redirection (e.g. 302 Moved Temporarily)
• 4xx Request Failure (e.g. 404 Not Found, 482 Loop Detected)
• 5xx Server Failure (e.g. 501 Not Implemented)
• 6xx Global Failure (e.g. 603 Decline)
SIP Signaling
SIP & H.323 Comparison
•
SIP is largely equivalent to the Q.931 and H.225 components of H.323 (sipcenter.com)
SIP
H.323
PHILOSOPHY
"New World" - a relative of Internet
protocols - simple, open and horizontal
"Old World" - complex, deterministic and vertical
IETF
ITU
Carrier-class solution addressing the wide
area
Borne of the LAN - focusing on enterprise
conferencing priorities
CHARACTERISTICS
A simple toolkit upon which smart clients
and applications can be built. It re-uses
Net elements (URLs, MIME and DNS)
H.323 specifies everything including the codec for the
media and how you carry the packets in RTP
Leaves issues of reliability to underlying
network
Assumes fallibility of network - an unnecessary
overhead
SIP messages are formatted as text. (Text
processing lies behind the web and email)
Binary format doesn't sit well with the internet - this
adds complexity
SIP allows for standards-based extensions
to perform specific functions.
Extensions are added by using vendor-specific nonstandard elements
Hierarchical URL style addressing scheme
that scales
Addressing scheme doesn't scale well
Minimal delay - simplified signalling
scheme makes it faster
Possibilities of delay (up to 7 or 8 seconds!)
Slim and Pragmatic
The suite is too cumbersome to deploy easily
SIP & H.323 Comparison
SIP
H.323
SERVICES
Standard IP Centrex services
Standard IP Centrex services
Ability to 'fork' calls
Not possible in the existing standard
User profiling
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'Unified messaging'
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Presence management
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Unique ability to mix media (e.g. IVR)
Cannot mix media within a session
URLs can be embedded in web browsers and
email tools
H.323 has no URL format
Works smoothly with media gateway controllers
controlling multiple gateways - crucial in a
multi-operator environment
"Shoehorn" interworking with SS7 is problematic - H.323
has trouble connecting calls to and from PSTN endpoints
Seamless interaction with other media services are only limited by the developers
imagination
Services are nailed-down and constricted - voice only ceiling
STATUS
Industry endorsed
Popularity due to the fact that it was the first set of agreedupon standards
Many vendors developing products
The majority of existing IP telephony products rely on the
H.323 suite
Security
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Integrity
Authenticity
Encryption
Intellectual rights protection
– Digital watermarking techniques embed extra
information into multimedia data
– Imperceptible to normal user and irremovable
Security
• At the IP layer security can be provided by
IPSec
• Secure RTP (RFC 3711)
– Provides 128 bit AES encryption
– Confidentiality, authentication and replay
protection
– SHA-1 (Secure Hash Algorithm) for
authentication
• Does not deal with key exchange