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Title: Paxos made simple


1
CS234/NetSys210 Advanced Topics in
Networking Spring 2012 SIP and VoIP Kundan
Singh, and Henning Schulzrinne "Peer-to-peer
internet telephony using SIP" in NOSSDAV
'05 Presentation by Swaroop Kashyap Tiptur
Srinivasa Anirudh Ramesh Iyer Tameem Anwar
2
Introduction to Session Initiation Protocol
(SIP)
3
Introduction to SIP
  • What is SIP ?
  • Text-based protocol (Defined in RFC 3261)
  • SIP Applications
  • Network Elements
  • User Agent (UAC and UAS)
  • Proxy Server (UAC and UAS)
  • Registrar
  • Redirect Server
  • SBC

4
SIP Messages
  • Request messages
  • REGISTER
  • INVITE
  • ACK
  • CANCEL
  • BYE
  • OPTIONS

5
SIP Messages
  • Response messages
  • PROVISIONAL (1XX)
  • SUCCESS (2XX)
  • REDIRECTION (3XX)
  • CLIENT ERROR (4XX)
  • SERVER ERROR (5XX)
  • GLOBAL FAILURES (6XX)

6
Assessment of VoIP quality over Internet
backbones Athina P. Markopoulou, Fouad A. Tobagi,
Mansour J. Karam
  • Quality of service for VoIP traffic in Internet
    backbone was studied based on delay as the
    metric.
  • 7 ISPs, 43 paths formed the test setup.
  • A model is proposed to study and analyze VoIP
    traffic in Internet backbones based on
    characteristics of VoIP such as talkspurts and
    silence periods.
  • This model is applied to the data trace obtained
    from the test setup (2.5 days worth of data).
  • The study shows that certain backbone paths are
    not equipped to handle VoIP traffic. Examples of
    such paths are coast-to-coast paths which have
    high delay.
  • The authors suggest that the primary reason for
    high delay in paths is due to the fact that there
    is no distinction between data and voice traffic.
    This prompts the authors to suggest IP QoS
    mechanism as a solution.

7
Improving VoIP Quality thorugh Path Switching Shu
Tao , Kuai Xu, Antonio Estepa, Teng Fei ,Lixin
Gao ,Roch Guerin, Jim Kurose, Don Towsley,
Zhi-Li Zhang
  • The effectiveness and benefits of path switching
    was examined, and its feasibility was
    demonstrated with the help of a of a prototype
    application-driven path switching gateway
  • With sufficient path diversity, path switching is
    indeed capable of yielding meaningful
    improvements in voice quality
  • The experiments also highlighted the benefit of
    adaptive decisions, especially in light of the
    often changing nature of the time scale at which
    network congestion takes place.
  • The study suggests that by exploiting the
    inherent path diversity of the Internet,
    application-driven path switching is a viable
    option in providing quality-of-service to
    applications
  • There is ongoing research being done to pursue
    these issues further in the context of hybrid
    wired/wireless networks and other applications
    such as video.

8
QoS-Enabled Voice support in the Next-Generation
Internet Issues, Existing Approaches and
Challenges Bo Li, Mounir Hamdi, Dongyi Jiang,
and Xi-Ren Cao, Hong Kong University of Science,
Technology ,Y. Thomas Hou, Fujitsu Laboratories
of America
  • There has been significant work done to establish
    the foundation to support VoIP. However, much
    remains to be done in order to ensure the QoS for
    VoIP and for multimedia traffic in general.
  • This article surveys the existing technologies
    to support VoIP, in particular the basic
    mechanisms in the IETF Internet telephony
    architecture and ITU-T H.323-related
    recommendations.
  • It then reviews the IETF QoS framework and major
    components in providing such QoS guarantees,
    including the Intserv and Diffserv models.
  • In addition, this article also presents two
    leading companies (Cisco and Lucent) solutions to
    offering IP telephony services
  • One another major issue currently under active
    development is internetworking with legacy net-
    works (i.e., PSTN). There are a number of
    proposals within the IEFT, in particular
    Media Gateway Control Protocol (MGCP).

9
Peer-to-peer internet telephony using
SIP Kundan Singh, and Henning Schulzrinne,
NOSSDAV '05
10
Peer-to-Peer Internet Telephony using SIP
  • SIP using Client-Server model
  • Less Robustness and Scalability
  • Increased costs due to Maintenance and
    Configuration
  • SIP using Peer-to-Peer model
  • Increased Robustness and Scalability
  • No maintenance and Configuration
  • Interoperability
  • Tradeoff
  • Resource look-up
  • Security

11
Distributed Hash Table ( DHT)
  • Types of Search
  • Central Index (Napster)
  • Distributed Index with flooding (Gnutella)
  • Distributed Index with hashing (Chord)
  • Basic Operations
  • find(key),insert(key , value),delete (key),
  • But no search

12
Background and Related work
  • Chord
  • Ring based Distributed Hash Table for structured
    P2P systems
  • Identifier Circle
  • Keys assigned to successor
  • Evenly distributed keys and nodes
  • Finger table- O(log N) entries
  • ith finger points to first node that
    succeeds n by at least 2i-1
  • Stabilization for join/leave
  • Iterative and Recursive Routing

13
Background and Related work
  • Skype
  • Based on Kazaa architecture
  • Open source P2P application for Internet
    telephony and instant messaging
  • Uses the concept of Super nodes.
  • Proprietary with Global Index server assigning
    Super nodes
  • Lookup similar to Kazaa using flooding unlike
    DHT-based lookup
  • Explicit server configurations not required

14
Background and Related work
  • P2P with SIP
  • Recent work have concentrated on combining SIP
    and P2P
  • SIP combination with P2P done in two ways
  • Replace SIP user registration and lookup by an
    existing P2P protocol
  • Implementation a P2P algorithm using SIP
    messaging
  • The paper takes the second approach
  • No modification done to SIP messages
  • Advantages- Use of existing SIP components
  • Disadvantages- Transport message size overhead

15
Difference with File Sharing
  • A single P2P-SIP node can handle many more
    requests than a file sharing node due to low data
    volume
  • Caching of location information is not useful.
  • The file sharing and directory lookup-based
    systems
  • can tolerate high lookup latency.
  • For file sharing applications, multiple almost
    exact copies of a popular file may be available.
    So node reliability does not matter.

16
Architecture
  • Server Farm architecture
  • P2P Overlay architecture
  • Hybrid Super-Node architecture

17
Server Farm architecture
  • Preserves Client-Server model
  • DHT can used in a Server farm
  • User registration done on only O(logN) servers
  • Redundancy in servers can prove expensive

18
Client Overlay architecture
  • Pure P2P overlay with all clients acting as a
    server
  • No server maintenance and configuration needed
  • Problem- Equal capacity and availability
  • Example- Client behind firewall or NAT

19
Super Node architecture
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  • Hybrid design (Similar to Kazaa)
  • Selection of Super nodes
  • High capacity (Bandwidth, CPU, memory)
  • Availability (up time, public IP address)
  • Transition from a regular node to Super node
    (Local decision)



20
P2P-SIP node block diagram
  • Discover-gtUser Interface-gtDHT(Chord)

21
Design and Implementation
  • Naming
  • Authentication
  • SIP messages
  • DHT discovery and join
  • SIP message routing
  • Reliability
  • Adaptor for existing SIP phones

22
Naming
  • Representation of end points using SIP URI's
  • Eg- sip17_at_192.1.2.38054
  • 17 is the key returned by Chord's hash function
  • Domain representation encouraged
  • sip10_at_example.com
  • SIP user identifiers (UI) randomly assigned
  • Authentication ?

23
Authentication
  • Validity of UI done when an user signs up to the
    P2P-SIP network
  • Absence of Public Key Infrastructure ?
  • Password obtained used in REGISTER authentication
  • Time-to-Live value used to determine user
    existence

24
SIP messages
  • SIP REGISTER- Used for registration and DHT
    maintenance
  • SIP REGISTER used in query and update mode
  • Query mode- User is asking for Contact
    information of the node identifier given in TO
    header
  • Update mode- Contact information present in
    message
  • Mostly done to update bindings given in TO header

25
DHT Discovery and Join
  • SIP REGISTER message used, Request-URI could be
    sip224.0.1.75
  • TO header identifies and discovers other P2P-SIP
    peers in the network
  • Once the node is discovered, SIP REGISTER is used
    to join the DHT
  • Chord Stabilization ? (Hint Registration of end
    points)

26
SIP message routing
  • Every node owns the responsibility for its subset
    of the key space
  • Destination key is determined from TO header
  • If Destination key belongs to same key space of
    the received node, then node is the registrar
  • Redirect or the proxy server is used to determine
    the locations available for the destination user.
  • Otherwise, request is proxied to the next hop
    node based on Chord algorithm and its inherent
    data structures

27
Reliability
  • Each Chord node stores log(N) successor addresses
  • User registrations replicated at K successive
    nodes
  • Users unregistered when exit is graceful
  • Registrar Node malicious ?

28
Advanced Services
  • Apart from user registration and call routing,
    which are the core services of SIP, P2P-SIP also
    offers some advanced services.
  • These services are based on SIP URIs. Example-
    sipstaff-meet_at_office.com or dialog.voicexml_at_ivr.
    net
  • The services are
  • Offline messages
  • Multi-party conferencing
  • NAT and firewall traversal
  • Directory Service

29
Advanced Services Offline Messaging
  • Offline messages Caller calls callee and callee
    is unavailable then a message is left by caller
    to callee. Needs storage and signaling.
  • Current P2P file storage IP telephony cannot
    handle this as we also need message waiting
    indication.
  • The P2P-SIP client running on the destination
    users machine can store the message if the
    destination user did not pick up the phone. The
    problem comes when the destination phone itself
    is not active or the user has not started her
    client.
  • The message waiting signal is implemented using
    POST (POST is a cooperative, decentralized
    messaging system that supports traditional
    services like email, news, IM) which is built on
    a P2P overlay.
  • To receive the offline message, the destination
    node subscribes to the message waiting indication
    (MWI) event with the P2P network and gets
    notified on startup when a new offline message is
    available .

30
Advanced Services Offline Messaging Contd
  • There are three places where we can store the
    of?ine messages the source, the destination or
    some intermediate node in the P2P overlay.
  • When Alice calls Bob and Bob is not online, the
    message should be stored reliably by the system
    and delivered to Bob when he comes online. Then a
    possible solution is to use the DHT peer
    responsible for storing Bob' location also store
    his offline messages.
  • In case of storage node failure message is
    replicated and kept consistent using Oceanstore
    architecture (a well-known architecture for
    global-scale persistent storage).

31
Advanced Services Offline Messaging Contd
  • Alternate model is to store the message at the
    sender-end itself.
  • If the message is not delivered or the storage
    node fails, then the caller node finds the new
    storage node and records the message again
    without any user intervention.
  • On boot-up a node checks for any undelivered
    message from past boot cycle, and tries to
    re-send them upon bandwidth and CPU availability.
  • This model has a security flaw What if the
    sender-end is an Internet kiosk?
  • To overcome this problem A third-party storage
    server can take the ownership of sending the
    message and store the message.

32
Advanced Services Multi-party Conferencing
  • There are 3 methods to do a multi-party
    conference using P2P-SIP,
  • A participating member can become mixer for small
    scale ad-hoc conferencing.
  • Completely decentralized approach can be taken to
    exchange member information and full-mesh
    signaling is established.
  • Multicast media distribution tree model can be
    used if number of senders are less at any point
    of time.
  • In all these models there is a trade-off in terms
    of reliability (single point of failure in case
    of single mixer), complexity and bandwidth
    utilization.

33
Advanced Services NAT and Firewall Traversal
  • In an ideal world, ISPs and corporate system
    administrators should enable their NAT and
    firewall devices with SIP proxies or application
    level gateways.
  • There are two aspects to overcoming the problem
    posed by ISPs
  • Automatic detection of the type of NAT and
    firewall. Detection is done at the application
    startup when the node connects to a super-node.
  • Tunneling through the NAT and firewall devices
    for inbound or outbound messages. The node
    implements the Interactive Connectivity
    Establishment (ICE) algorithm for NAT traversal
    (Can use both TCP and UDP). Also every node has a
    built-in STUN (Simple Traversal of UDP through
    NAT) and TURN (Traversal Using Relay NAT) server.

34
Advanced Services Directory Service
  • In any chat application people tend to search by
    first-name or last-name. Usually only partial
    names are supplied, which leads to wildcard
    character behaviour.
  • Queries could also be with respect to number of
    hops from the person issuing the query.
  • General DHT does not support these type of
    queries.
  • These queries are supported by,
  • Registering combination of first-name and
    last-name in DHT.
  • Doing a blind search in acquaintances graph based
    on small number of hops-to-live.

35

Performance prediction - Scalability
  • Scalability
  • With N super-nodes and n nodes in the chord
    system, number of keys k per node n/N.
  • REGISTER refresh rate is rs, refresh rate for
    finger table entry is rf.
  • Call arrival (Possion distribution) with mean c,
    user registration (Uniform distribution) with
    mean t per user, Node churn (Possion
    distribution) with mean ?.
  • Average lookup time in Chord is O(log N), as
    there are O(log N) finger table entries Node join
    leave messages generated is O( (log N)2)
  • The average message rate per node is sum of the
    message rates due to refresh, call arrival, user
    registration and node join or leave
  • If each node can handle C requests per second,
    then the equation C M gives the maximum
    possible number of nodes, Nmax, in the system,
    which roughly translates to Nmax 2(C/(rc))
    for large N, where r is the refresh rate and c is
    the call rate. Example If a node supports 10
    requests per second, r1min, c1per min, then max
    nodes in system 2(1030).

36

Performance prediction - Reliability
  • Reliability
  • What happens when a node fails? Impacted entity
    is user registration records/data in that node.
  • Reliability is achieved by,
  • Refresh rate is increased so that node failure
    detection happens quickly.
  • User registration refresh rate is increased so
    that loss is very brief.
  • User registration record is replicated at
    multiple nodes at log(N) successive nodes.

37

Performance prediction Call setup latency
  • Latency
  • O(log N) nodes to be looked-up before a call can
    be setup. This is a bottleneck in reducing call
    setup latency.
  • In a 10000 node cluster it takes 6 hops 1-2
    seconds before call is made. This is higher than
    vanilla SIP (200ms) but can be tolerated as
    phones ring for a much higher time than 1-2
    seconds.
  • P2P synchronization latency due to churn in
    nodes/records can result in multiple
    retransmissions before call setup is complete.
    Example skype takes 3-8 seconds.
  • To solve this issue a hybrid model of structured
    unstructured P2P network is needed with one hop
    lookup. This solution assumes large storage space
    is available in peers.

38
Open Issues
  • Security, trust and reward
  • Security threats Trust, Malicious behaviour by
    nodes, DoS
  • Trust Proprietary protocols like Skype has
    built-in trust among peers but open protocols
    like P2P-SIP may have malicious clients.
  • Reward As in any P2P Reward nodes serving in
    DHT, punish free-riders.
  • Media routing
  • In case of media transfer paths we need media
    relay node in NAT and firewall scenarios.
    Stability of relay node is important as delay due
    to relay node dropping out is not tolerable.

39

Thank You !
40

Detailed slides of QoS papers
41
CS234/NetSys210 Advanced Topics in
Networking Spring 2012 Assessment of VoIP
quality over Internet backbones Athina P.
Markopoulou, Fouad A. Tobagi, Mansour J.
Karam Presentation by Swaroop Kashyap Tiptur
Srinivasa Anirudh Ramesh Iyer Tameem Anwar
42
Introduction to Voice over IP (VoIP)
43
Introduction to VoIP
  • What is VoIP ?
  • VoIP Protocols
  • MGCP Media Gateway Control Protocol
  • RVP over IP Remote Voice Protocol Over IP
    Specification
  • SGCP Simple Gateway Control Protocol
  • SIP Session Initiation Protocol
  • Skinny Skinny Client Control Protocol (SCCP)

44
VoIP Architecture
  • Coding Decoding of Analog Voice
  • Analog-to-Digital and Digital-to-Analog
    conversions
  • Compression
  • Signaling
  • Call setup tear down
  • Resource coding negotiation
  • Transport of Bearer Traffic
  • Voice packet transmission
  • Routing
  • Support of quality of service
  • Numbering
  • Phone number, IP address

45
VoIP Architecture - Contd
  • A naive representation of VoIP system as shown
    in this paper
  • Sender
  • Encoder Does sampling and creates a constant bit
    rate stream
  • Packetizer encapsulates speech samples into
    packets of equal sizes
  • Receiver
  • Playback buffer An important component which
    absorbs delay/jitter
  • De-packetizer Decoder Reconstruct the speech
    signal

46
VoIP Architecture - Contd
  • Codecs They are used to convert an analog voice
    signal to digitally encoded version and
    vice-versa.
  • Codecs vary in the sound quality, bandwidth
    required, computational requirements, etc.
  • Each service, program, phone, gateway, etc
    typically supports several different codecs, and
    when talking to each other, negotiate which codec
    they will use.

47
Assessment of VoIP
  • Goal of the paper Assessment of VoIP over
    today' Internet
  • Organization of the paper
  • Section I II describes the components of VoIP
    system under evaluation ? Already Covered in
    Architecture
  • Section III presents the quality measures used
    for assessing the impairments over the network
    and our methodology for rating a call
  • Section IV describes the probe measurements and
    classify the traces into categories according to
    their delay and loss characteristics
  • Section V application of proposed methodology to
    the traces discuss numerical results pertaining
    to phone calls quality
  • Section VI concludes the paper

48
Assessment of VoIP Quality Metrics
  • Delay and Loss
  • Mean Opinion Score (MOS)
  • It is a numerical method of expressing voice and
    video quality
  • MOS is quite subjective, as it is based figures
    that result from what is perceived by people
    during tests
  • MOS is expressed in one number, from 1 to 5, 1
    being the worst and 5 the best
  • Emodel
  • It is a computational model, standardized by
    ITU-T, that uses transmission parameters to
    predict the subjective quality of packetized
    voice

49
Assessment of VoIP Quality Metrics Contd
  • How do we obtain a subjective score (MOS) from
    metrics like delay, jitter, loss?
  • Emodel provides a solution to this problem by
    defining a computation model (based on
    statistical analysis)
  • First we calcluate R rating of a call using the
    formula, R (Ro - Is) - Id - Ie A, where Ro
    (effect of noise) and Is (accounting for loud
    connection and quantization) terms are intrinsic
    to the voice signal. Id and Ie capture the effect
    of delay and signal distortion respectively.
  • More about Id and Ie in next slide.

50
Assessment of VoIP Quality Metrics Contd
  • Delay impairment factor Id, models the quality
    degradation due to one-way or mouth-to-ear(m2e)
    delay.
  • Id Idte(m2e,EL2) Idle(m2e,EL1) Idd(m2e),
    where terms Idte(m2e,EL2) and Idle(m2e,EL1)
    capture the impairments due to talker and
    listener echo respectively. EL 8 (infinite echo
    loss) corresponds to perfect echo cancellation.
    Term Idd(m2e) captures the interactivity
    impairment when the m2e delay is large, even with
    perfect echo cancellation.
  • Along-with this definition a new parameter task
    is defined to measure interactive nature of call
    (eg business conference which consists of
    continuous bursts by different people OR normal
    Bob-Alice conversation). 1 is defined to be
    highly interactive (random and bursty), 6 is
    defined to be relaxed and free conversation.

51
Assessment of VoIP Quality Metrics Contd
  • Loss impairment Ie captures the distortion of the
    original voice signal due to low-rate codec, and
    packet loss in both the network and the playback
    buffer.
  • Table shows effect of Codec on this parameter.

52
Assessment of VoIP Quality Metrics Contd
  • Observed facts about Ie and Id in different calls
    (more about test setup in actual methodology and
    results slides)

53
Assessment of VoIP Quality Metrics Contd
  • Finally, now that we have R rating by evaluation
    Ie and Id we can get MOS by using the
    translation shown below. (Needed because PSTN
    call quality is measured in MOS and is gt3.6)

54
Assessment of VoIP Quality Metrics Contd
  • An important factor/issue while measuring the
    parameters that we have ssen so far is the issue
    of using correct time interval to measure these
    parameters.
  • For long duration calls a natural approach is to
    divide the call duration into fixed time
    intervals and assess the quality of each interval
    independently.
  • Evaluating each interval in terms of Ie leads to
    transitions between plateaus of quality. (Also
    accounts human memory about bad/good transitions
    by using distribution for forgetting time
    interval when calculating instantaneous MOS. Bad
    transition is high loss aka bursts and Good
    transition is low loss aka gaps)

55
Assessment of VoIP Internet Measurements
  • A model with adaptive playback delay is used in
    measurement to simulate real-VoIP implementations
    and experiment is conducted to measure the
    parameters involved in Call Quality.
  • Probes are configured across 7 different Service
    Providers in 5 major cities to measure the
    parameters for a duration of 2.5 days.
  • A probe packet (based on codec scheme and
    adaptive playback delay) consists of 50 Bytes of
    data sent every 10ms. 50Bytes 10Bytes of data
    40Bytes of RTP/TCP/IP headers).
  • This corresponds to G.729 rate of 8Kbps
  • A total of 43 paths are tested to assess their
    support for VoIP and create a model/rules to
    ensure Internet backbone can support toll-quality
    VoIP.

56
Assessment of VoIP Internet Measurements
  • The following facts could be observed from the
    probe/trace data
  • The paths could be classified into different
    types (A,B, C, D, E). Paths of type A and B
    connect ASH, EWR and AND on the east coast and
    have low propagation delays, i.e. below 10ms.
    Paths of type C, D on E on the other hand,
    connect cities across the US.
  • Based on the variable component of delay, paths
    of type A and C have practically no queuing.
    Paths of type B and D have in general low
    queuing, except for clustered delay spikes. Paths
    of type E are coast-to-coast loaded paths. The
    queuing component is high and the delay varies
    slowly in a short time scale.

57
Assessment of VoIP Internet Measurements
  • Only 3 out of the 43 paths had consistently no
    loss during the 2.5 days observed. The rest of
    them incurred loss durations that varied from
    10ms up to 33.72sec, although the average loss
    rateswere low (e.g. lt0.2).
  • 6 out of 7 providers experienced outage periods
    10-220sec for 1-2 times per day. These occurred
    due to convergence delay following route change.
  • 0.5-2sec loss durations were correlated with
    delay spikes.
  • The number of out-of-order packets was
    negligible.

58
Assessment of VoIP Results
  • So far we have covered Call Quality Metrics,
    Model to assess quality and Trace data
    obtained from probes
  • To tie it all together Model obtained from Call
    Quality Metrics is applied to Trace data to
    validate the model and understand parameters that
    drive VoIP quality

59
Assessment of VoIP Results
  • Evaluating the trace data using an example a 15
    minute call on path of type E , it can be seen
    that a fixed playout of 100ms ensures that Ie is
    low and optimal region is 200ms for fixed
    playout (MOS 4).
  • Even adaptive model has a playout close to it
    (122ms) with max MOS 3.6

60
Assessment of VoIP Results Contd
  • For the trace example in previous slide, the
    loss-delay tradeoff graph with variation in type
    of task and echo-loss looks like,

61
Assessment of VoIP Results Contd
  • Looking at instantaneous quality of calls
    initiated at random times spread over an hour, it
    can be seen that 150ms playback delay is optimal
    choice as only 6 have final MOS lt3.6
  • 100ms playback results in high percentage of
    calls in low MOS region.
  • Adaptive playback delay shows linear nature

62
Assessment of VoIP Results Contd
  • Looking at instantaneous quality of calls
    initiated at random times spread over an entire
    day, it can be seen that 150ms playback delay is
    optimal choice as only 6 have final MOS lt3.6
  • 100ms playback results in high percentage of
    calls in low MOS region.
  • And adaptive playback delay shows no particular
    performance benefit with 10 of calls having MOS
    lt 3.5

63
Assessment of VoIP Results Contd
  • The previous graphs/results were based on time
    duration on same path.
  • Impact of different types of paths are,
  • Paths of type A and B have low delay and low
    delay variability. A fixed playback delay of
    100ms is sufficient for MOS gt 4
  • Paths of type B and D exhibit periodically
    clusters of high spikes. Performance is low in
    both fixed adaptive delays. Only a very high
    fixed delay results in 10 of calls with MOS lt
    3.5

64
Assessment of VoIP Results Contd
  • Combining inferences drawn for different times
    and paths the following results are obtained,
  • Some ISP backbones (i.e. those that are
    over-provisioned and have low delay variability,
    namely types A and C) are indeed able to provide
    high quality VoIP today. This is true for both
    short and long distance paths.
  • Highly loaded paths (type E) as well as some
    over-provisioned paths exhibiting frequent delay
    spikes (types B and D) have poor VoIP
    performance. Under the best scenarios these paths
    are barely able to provide acceptable (MOS gt 3.6)
    VoIP service.
  • The poor VoIP performance on loaded backbones
    (type E) makes a strong case for separating voice
    and giving it priority. It also needs a more
    sophisticated handling than over-provisioning.
  • There exists a tradeoff between delay and buffer
    loss. Maximum MOS is more sensitive to an
    increase in loss rather than to an increase in
    delay. The reason for this is that the underlying
    Ie curves are sharper than the Id curves.

65
Assessment of VoIP Results Contd
  • Combining inferences drawn for different times
    and paths the following results are obtained,
  • The need for adaptive playout comes when EITHER
    the delay is high and there is no margin for
    overestimating it OR, when the delay is unknown
    and the receiver does not know how to select an
    appropriate fixed value.
  • Tuning of parameters such as thresholds for spike
    detection, weights used for moving averages
    cannot be realistically optimized (proven with
    observations).
  • Delay estimation mechanism has weakness of using
    TCP-like prediction (p d4v) which tends to
    over-estimate delays beyond what is appropriate
    to preserve interactivity. Also adapting at the
    beginning of talkspurts fails to react to short
    lived spikes while it still unnecessarily leads
    to high delays.

66
Assessment of VoIP Conclusion
  • After seeing the results and considering,
  • Delay and loss
  • Voice quality
  • Playback buffer schemes
  • Conclusion
  • Many paths performed poorly for VoIP traffic
    due to long delays and large delay variability
  • Playback buffer schemes tradeoff between data
    loss and increased delay in the buffer
  • Overall rated VoIP as poor
  • Mark voice traffic and give it preferential
    treatment
  • Choose playback buffer scheme to match delay
    pattern


67

Questions?
68
CS234/NetSys210 Advanced Topics in
Networking Spring 2012
Improving VoIP Quality thorugh Path Switching Shu
Tao , Kuai Xu, Antonio Estepa, Teng Fei ,Lixin
Gao ,Roch Guerin, Jim Kurose, Don Towsley,
Zhi-Li Zhang
Presentation by Swaroop Kashyap Tiptur
Srinivasa Anirudh Ramesh Iyer Tameem Anwar
69
  • Introduction
  • i)VoIP
  • ii) Factors affecting VoIP Quality
  • Architecture of Path Switching System
  • How Path Quality is Estimated
  • Issues about Path Switching
  • Performance Evaluation

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VoIP requires minimum service guarantees that go
beyond the best-effort structure of todays IP
networks
72
  • 1) Network Factors
  • Packet Loss
  • Delay Jitter
  • Network Delay
  • 2) Application Factors
  • Playout Buffers
  • Codec Performance

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APS Gateway Components
  • Path Prober
  • Send UDP probes , which are generated to emulate
    the behavior of a VoIP call, on each available
    path periodically
  • Receive feedbacks containing the delay and loss
    statistics.
  • Path Quality Estimator
  • Translate network path measurements into
    application quality estimates using the E-model

76
APS Gateway Components (Cont)
  • Path Selector
  • Dynamically decide the best performing path for
    each voice call
  • Packet Forwarder
  • Integrated with the path selector
  • Forward the voice packet along the corresponding
    path.

77
  • Perceived voice quality is typically measured by
    the Mean Opinion Score (MOS), a subjective
    quality score that ranges from 1 ( unacceptable)
    to 5 (excellent).
  • The MOS method is from Methods for subjective
    determination of transmission quality , ITU-T
    Recommendation P.800, August 1996.

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The ITU-T E-Model defines a R-factor ,that ranges
from 0 to 100 ,combines different aspects of
voice quality impairments The E-Model method
is from The E-Model , a computational model for
use in transition planning , ITU-T
Recommendation G.107, March 2003
80
  • Among all of the factors in Equation 1 ,only Id
    and Ie are typically considered variable in a
    VoIP System . Using default values for all other
    factors reduces the model to
  • R 94.2 Ie - Id
  • When R94.2 , the value of MOS is 4.4

81
  • Ie accounts for impairment caused by both
    encoding and transmission losses

82
  • Id accounts for impairment caused by network
    delay, codec-related delay and play out buffering
    delay.

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  • Path Quality Prediction
  • Estimating the Benefits of Path Switching
  • Time- Scale Adaptive Path Switching Algorithm

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A typical example that shows how path switching
avoids quality degradations (a) quality
variations on the two paths (top), and (b) the
resulting quality when path switching is applied
(bottom).
88
Conclusion
  • The effectiveness and benefits of path switching
    was examined, and its feasibility was
    demonstrated with the help of a of a prototype
    application-driven path switching gateway
  • With sufficient path diversity, path switching is
    indeed capable of yielding meaningful
    improvements in voice quality
  • The experiments also highlighted the benefit of
    adaptive decisions, especially in light of the
    often changing nature of the time scale at which
    network congestion takes place.

89
  • The study suggests that by exploiting the
    inherent path diversity of the Internet,
    application-driven path switching is a viable
    option in providing quality-of-service to
    applications
  • Intelligently creating and exploiting path
    diversity via mechanisms such as overlays and
    dynamic path switching is therefore an important
    avenue to meet and improve the quality-of-service
    of applications
  • There is ongoing research being done to pursue
    these issues further in the context of hybrid
    wired/wireless networks and other applications
    such as video.

90
The APS gateways
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CS234/NetSys210 Advanced Topics in
Networking Spring 2012
QoS-Enabled Voice support in the Next-Generation
Internet Issues, Existing Approaches and
Challenges Bo Li, Mounir Hamdi, Dongyi Jiang,
and Xi-Ren Cao, Hong Kong University of Science,
Technology ,Y. Thomas Hou, Fujitsu Laboratories
of America
Presentation by Swaroop Kashyap Tiptur
Srinivasa Anirudh Ramesh Iyer Tameem Anwar
93
Introduction
  • While the Internet has served as a research and
    education vehicle for more than two decades, the
    last few years have witnessed its tremendous
    growth and its great potential for providing a
    wide variety of services
  • Over the past few years, reliability and quality
    have quickly improved, and Internet telephony is
    now one of the fastest growing industries.
  • The reason behind Internet telephonys success
    is that it can potentially bring enormous
    benefits to end users, telecommunication
    companies, and carriers.

94
Advantages of IP Telephony
  • It is cheaper for end users to make an Inter-
    net telephony call than a circuit-switched call,
    mainly because operators can avoid paying
    interconnect charges.
  • Internet telephony gives new operators an easy
    and cost-efficient way to compete with incumbent
    operators
  • Engineering economics favors Internet telephony.
    While a circuit-switched telephony call takes up
    to 64 kb/s, an Internet telephony call only takes
    up to 68 kb/s and possibly even less bandwidth.

95
Advantages of IP Telephony(Cont.)
  • In the longer term, it offers exciting new
    value-added opportunities such as high fidelity
    stereo conferencing bridges, Internet multicast
    conferencing, and telephony distance learning
    applications, phone directories and screen
    popping via IP, or even voice Web browsing.
  • Internet telephony gives carriers the ability to
    manage a single network handling both voice and
    data. Internet telephony will also create end
    user opportunities and demand for new services.

96
Latency
  • Latency has been constantly undergoing changes
    and will continue to improve, driven by three
    factors
  • Improved gateways (developers are just
    beginning to squeeze latency out of the first
    generation of products)
  • Deployment over private networks by deploying
    gateways on private circuits, organizations
  • Service providers can control the bandwidth
    utilization and hence latency

97
Objective
  • The objective of this article is to review the
    recent developments and key enabling
    technologies in providing QoS supporting for
    voice communications in the next-generation
    Internet. The rest of the article is organized as
    follows. We first review the existing
    technologies in supporting VoIP networks,
    especially the basic mechanisms in the IETF
    Internet telephony architecture.

98
Internet Telephony Standards
  • To support Internet telephony and other related
    applications, standards are being recommended
    and developed to insure interoperability
  • ITU H.323 specification for Internet telephony
    is gaining widespread acceptance among software
    vendors
  • The IETF is developing protocols such as Session
    Initiation Protocol (SIP) for multimedia
    session initiation, and RTSP for controlling
    multimedia servers on the Internet that can work
    together with H.323.

99
Internet Telephony Standards(cont)
  • Real-Time Transport Protocol (RTP) is
    interwoven with all the above protocols. It is
    used by H.323 terminals as the transport
    protocol for multimedia both SIP and RTSP were
    designed to control multimedia sessions
    delivered over RTP.
  • To this end, it guarantees that each participant
    in a session has a unique identifier, providing
    applications a way to de-multiplex packets from
    different users.
  • RTP also contains a control component, called
    the Real-Time Control Protocol (RTCP). It is
    multicast to the same multicast group as RTP,
    but on a different port number.

100
Internet Telephony Standards(cont)
  • One of the key components supporting VoIP is a
    signaling protocol, which has to provide the
    following functions
  • User location
  • Session establishment
  • Session negotiation
  • Call participant management
  • Feature invocation
  • Within the IETF, two protocols are defined
    to implement these tasks Session Initiation
    Protocol(SIP) and Session Description Protocol
    (SDP)

101
Session Initiation Protocol
  • SIP is used to initiate a session between
    users. It provides user location services, call
    establishment, call participant management, and
    limited feature invocation.
  • SIP is a client- server protocol. This means
    that requests are generated by one entity
    (client), and sent to a receiving entity (the
    server), which process them.
  • There are three types of servers. SIP requests
    can traverse many proxy servers, each of which
    receives a request and forwards to the next-
    hop server, which may be another proxy server or
    the final user agency server. A server may also
    act as a redirect server, informing the
    client of the next-hop server so that the client
    can contact it directly.

102
Session Description Protocol
  • SDP is used to describe multimedia sessions
    for both telephony and distributed applications.
    The protocol includes several kinds of
    information, as follows
  • Media streams convey the type for each media
    stream. For each media stream, the destination
    address (unicast or multicast ) is indicated
    by Address.
  • Ports define the UDP port numbers for each
    sending or/and receiving stream.
  • Payload type conveys the media formats that can
    be used during the session.
  • For a broadcast-style session such as a
    television program, start and stop times convey
    the start, stop, and repeat times of the
    session
  • Originator names the originator of the session
    and how that person can be contacted.

103
Basic Mechanisms in H.323
  • H.323 are a series of Recommendations of the
    ITU-T to enable multimedia communications in
    packet switched networks H.323 is designed to
    extend the traditionally circuit-based services
    including audiovisual and multimedia conferencing
    services into packet-based networks
  • One of the primary objectives of H.323 is the
    interoperability with the existing
    circuit-switching systems (PSTN and ISDN).
  • The basic elements defined in H.323
    architecture are terminals, gateways,
    gatekeepers, and multipoint control units (MCUs),
    in which the terminals, gateways, and MCUs are
    collectively referred as endpoints.

104

The H.323 Protocol Stack
105
Basic Mechanisms in H.323(cont)
  • A gateway, as the name suggests, is an
    intermediate device to provide interoperation
    between H.323 compliant devices and non-H.323
    devices, in particular PSTN and ISDN devices.
  • A gatekeeper manages a set of registered
    endpoints, collectively referred as a zone. Its
    main functions include call admission (or call
    authorization), address resolution, and other
    management-related functions.
  • An MCU provides the necessary control needed
    for multiparty video conferences. It contains
    two logical components a multipoint controller
    (MC) for call control coordination and a
    multipoint processor (MP) to handle audio or
    video mixing.

106
H.323 Protocol Phases
107
The IETF Differentiated Services Framework
  • The Diffserv architecture is based on a simple
    model where traffic entering a network is
    classified and possibly conditioned at the
    boundaries of the network, and assigned to
    different behavior aggregates (BAs), with each
    BA being identified by a single Diffserv code-
    point (DSCP)
  • Sophisticated classification, marking, policing,
    and shaping operations need only be implemented
    at network boundaries or hosts .
  • A Diffserv architecture can be specified by
    defining or implementing the following four
    components
  • The services provided to a traffic aggregate
  • The traffic conditioning functions and PHBs used
    to realize the services
  • The Diffserv field value (DSCP) used to mark
    packets to select a PHB
  • The particular node mechanism to realize a PHB.

108
There are two approaches to provide Diffserv
The first approach specifies the QoS in
deterministically or statistically quantitative
terms of throughput, delay, jitter, and/or loss.
Such approach is called quantitative Diffserv.
The second approach specifies the services in
terms of some relative priority of access to
network resources and is called priority based
Diffserv.
109
The CISCO Solution Enterprise IP Telephony
  • The Cisco solution for IP telephony in
    enterprise networks includes hardware, such as
    switches, routers, IP/PSTN gateways, desktop IP
    phones, and software, such as the call manager
  • By using routers and gateways to connect the
    PBX, voice traffic can be carried over data IP
    networks. Call management soft- ware and IP
    telephones are deployed in the existing IP
    networks at each remote site. This will reduce
    the cost of WAN consolidation while at the same
    time eliminating the cost of installing a second
    network at each remote location.
  • Packet classification identifies and cate-
    gorizes network traffic into multiple classes.
    The Cisco IP phone can set the IPv4 ToS at the
    ingress to the network.

110
The CISCO Solution Enterprise IP Telephony
  • The QoS guarantees are primarily provided by
    two mechanisms
  • The call manager equipped with a resource
    reservation protocol (e.g., RSVP)
  • A priority queue mechanism.
  • The priority queue mechanism is maintained
    in the core routers, and is responsible for
    high-speedswitching and transport as well as
    congestion avoidance.

111
The Cisco Data and IP telephony configuration
112
Lucent Gateway Solution ForService Provider
Networks
  • In this architecture an H.323 or SIP-compliant
    terminal is connected to the IP switch or
    router. The edge switches or routers serve as
    access points and concentrators for the core IP
    network, which comprises higher-capacity IP
    routers or switches.
  • Two gateways are added to the IP network
    architecture as interfaces to the PSTN. The
    first added is a connection gateway (CG), which
    performs signaling interworking between the IP
    protocol and PSTN protocols. The second is a
    voice gateway (VG), which converts time division
    multiplexed signals into IP packet and vice versa

113
Lucent IP and PSTN Architecture
114
Difference between Lucent and Cisco Solutions
  • The Lucent router implements a straightforward
    scheme for QoS. It simply extracts ToS
    information from incoming IP packets and sets up
    a series of prioritized queues. These queues
    can control packet flow based on the CoS value,
    which allows the router to prioritize voice
    data and move fax data to a lower priority,
    thereby minimizing delay on real-time information
    at the expense of less time-critical
    information.
  • The difference between these two approaches lies
    in the fact that the Cisco system is targeted
    for the enterprise network, in which per flow
    end-to- end QoS guarantee is possible
  • The Lucent approach is used for carrier
    networks, which is more scalable but relies on
    the underlying IP network to provide the needed
    QoS.

115
Conclusion
  • There has been significant work done to establish
    the foundation to support VoIP. However, much
    remains to be done in order to ensure the QoS for
    VoIP and for multimedia traffic in general.
  • This article surveys the existing technologies
    to support VoIP, in particular the basic
    mechanisms in the IETF Internet telephony
    architecture and ITU-T H.323-related
    recommendations.
  • It then reviews the IETF QoS framework and major
    components in providing such QoS guarantees,
    including the Intserv and Diffserv models.
  • In addition, this article also presents two
    leading companies (Cisco and Lucent) solutions to
    offering IP telephony services
  • One another major issue currently under active
    development is internetworking with legacy net-
    works (i.e., PSTN). There are a number of
    proposals within the IEFT, in particular
    Media Gateway Control Protocol (MGCP).

116
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