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VoIP over the WiMAX

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VoIP over the WiMAX Adviser: Ho-Ting Wu Presenter: Chi-Fon Yang Outline VOIP Protocols IEEE 802.16 Introduction Voice over Ethernet via IEEE 802.16 QoS Strategy for ... – PowerPoint PPT presentation

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Title: VoIP over the WiMAX


1
VoIP over the WiMAX
  • Adviser Ho-Ting Wu
  • Presenter Chi-Fon Yang

2
Outline
  • VOIP Protocols
  • IEEE 802.16 Introduction
  • Voice over Ethernet via IEEE 802.16
  • QoS Strategy for VoIP Services over IEEE 802.16
  • Conclusions
  • References

3
Outline
  • VOIP Protocols
  • Uses the Internet Protocol (IP) to transmit voice
    as packets over an IP network.
  • The main focus is on H.323 and SIP (Session
    Initiation Protocol) .

NEWS
4
NEWS
  • Airspan Networks??,?870?????VOIP???????????ArelNet
    ?? ,ArelNet??????????VOIP???????,????????WIMAX????
    IP?????
  • ????TW-WiMAX???? ?2006????????????,???????????????
    ?????WiMAX?? ,????????????????????????????????????
    ?????? ?
  • ??????????????????????????
  • ????????-???
  • ??????????????????,?????????????,?????PSTN?????VoI
    P(????)??.
  • ???????ISP?? - NextWeb.????????WiMAX?????VoIP??

BACK
5
VOIP Protocols
  • Signaling protocols
  • H.323 , SIP are used to setup the route for the
    transmission over the IP network
  • Signaling transport (SIGTRAN) are used transport
    SS7 over IP
  • Gateway protocols
  • Media Gateway Control Protocol (MGCP/MEGACO) are
    used to establish control and status in the media
    and signaling gateways.
  • Routing (UDP,TCP) and transport protocols (RTP)
    are used once the route is established for the
    transport of the data stream

6
H.323
  • ITU-T standard
  • Provides the technical requirements for voice
    communication over IP service
  • Provides audio, video and data communications
    across IP-based networks.
  • Control protocols
  • H.225.0/Q.931 Call Signaling
  • H.225.0 RAS
  • H.225.0 Call Signaling
  • H.245 Media Control

7
Components of H.323
  • Terminal
  • The LAN client endpoints that provide real time
  • Support H.245, Q.931, RAS, RTP,MCU
  • Gateways
  • Two-way communications between H.323 terminals
  • Translation between different transmission
    formats (e.g from H.225 to H.221 )
  • Translating between audio and video codecs
  • Gatekeepers
  • Address Translation
  • Admissions Control
  • Call signaling
  • Bandwidth Management
  • Call Management
  • Multipoint Control Units (MCU)
  • Provides the capability for three or more
    terminals and gateways to participate in a
    multipoint conference

Call-Setup
8
Call Setup in H.323
Gatekeeper
Terminal
Terminal
Back
9
SIP - Session Initiation Protocol
  • IETF standard for VOIP
  • RFC 3261 from the Internet Engineering Task Force
    (IETF)
  • Application layer control protocol for creating,
    modifying and terminating sessions
  • Similar to that of HTTP or SMTP
  • Text-encoded protocol
  • Main functions
  • ?Invite users to sessions
  • Find the users current location, match with
    their capabilities and preferences
  • ?Modification of sessions
  • ?Termination of sessionsText-based Encoding

10
SIP Architecture
SIP Request
SIP Response
RTP Media Stream
Redirect Server
Location Server
Proxy Server
Proxy Server
Proxy Server
11
Invitation for SIP Proxy-
itri.org.tw
location server
csie.nctu.edu.tw
acer_at_csie.nctu.edu.tw
AUDI
BMW
BENS
Honda_at_AUDI
Back
12
Invitation for Redirect Server-
location server
ccl.itri.org.tw
acer_at_ccl.itri.org.tw
BENS
BMW
AUDI
Back
13
Invitation for Registration Server-
Back
REGISTER F1
client_at_user.com
SIP Server
John
200 OK F2
user.com
register.com
14
COMPARISON OF H.323 WITH SIP
COMPARISON OF H.323 WITH SIP COMPARISON OF H.323 WITH SIP COMPARISON OF H.323 WITH SIP
Protocol H.323 SIP
Standard ITU IETF
Message Definition ASN.1 ABNF (RFC2234)
Message Encoding compact binary format UTF-8 text format
Scalability Gatekeepe r( Not very scalable ) Proxy (Highly scalable )
Authentication H.235 HTTP RFC2617
Encryption H.235 PGP, SSL
Call Setup Delay 7 RTT 1.5 RTT
Media Transport RTP/RTCP RTP/RTCP
Transport Protocol TCP/UDP TCP/UDP/STCP
Conference MC/MCU Media Server
Hesder Field Binary representation Textual representation
15
Media Gateway Control Protocol MGCP/MEGACO
  • Defines communication between call control
    elements (Call Agents) and telephony gateways.

16
MGCP/MEGACO Introduction
  • A protocol for controlling media gateways
  • Components
  • Media Gateway (MG)
  • provides conversion between audio signals on
    telephone circuits and data packets carried over
    IP networks.
  • Trunking gateway
  • to CO/switches
  • Residential gateway
  • to PSTN phones
  • Media gateway controller (MGC)
  • handles the call setup and release for media
    channels in a media gateway.

Call-Setup
17
MGCP Call Setup
Back
18
Softswitch
MG
19
IP Media Subsystem Services - IMS
  • Today WLAN is the only standard capable of
    providing the Quadruple play Technologies- Data,
    Voice, Video, Mobility (802.16e) using a single
    network
  • Comprising all Core Network elements for
    provision of multi media services
  • Utilizes the Packet Switch domain to transport
    multimedia signaling and bearer traffic (SGSN
    GGSN)).

Back
20
Interworking with PSTN
  • Call State Control Function (CSCF)
  • Incoming Call Gateway (ICG)
  • Acting as a first entry point to perform routing
    of incoming calls
  • Call Control Function (CCF)
  • Call setup/termination and call state/event
    management
  • Serving Profile Database (SPD)
  • Interacting with HSS to receive user profile
    information
  • Address Handling (AH)
  • Mapping between alias address(e.g.,E.164number)and
    IP address
  • Roaming Signaling Gateway (R-SGW) -
  • Providing communication with a 2G/R99MSC/VLR
  • Multimedia Resource Function (MRF)
  • Performing multi-party call and multi-media
    conferencing functions
  • The same function as an MCU in theH.323 network

2.5G/3G
  • Media Gateway Control Function (MGCF) -
  • Performing protocol conversion between the legacy
    (e.g.,ISUP) and the All-IP network call control
    protocols
  • Home Subscriber Server (HSS) -
  • Master database for a given user
  • Functionalities
  • The circuit switched part of the HLR
  • The HLR functionality required by the PS-Domain
  • User control functions required by the IMS
  • Transport Signaling Gateway (T-SGW) -
  • Mapping call related signaling from PSTN/PLMN on
    an IP bearer and sending it to the MGCF
  • Providing PSTN/PLMN IP transport level address
    mapping

Back
21
Outline
  • VOIP Protocols
  • IEEE 802.16 Introduction

22
IEEE 802.16 Introduction
  • FEATURES OF WIMAX
  • Scalability
  • The standard supports hundreds or even thousands
    of users within one RF channel
  • As the number of subscribers grow the spectrum
    can be reallocated with process of sectoring
  • Quality of Service
  • Range
  • Optimized for up to 50 Km
  • Designed to handle many users spread out over
    kilometres
  • Coverage
  • Standard supports mesh network topology
  • Downlink Channel
  • Downlink Channel Descriptor (DCD)
  • Transmitted by the BS
  • Define the characteristics of a downlink physical
    channel
  • Downlink Map
  • PHY Synchronization
  • DCD Count
  • Match the DCD
  • Base Station ID
  • 48-bit
  • SBC-REQ / SBC-RSP management message
  • SBC-REQ stands for SS Basic Capabilities
    Request.
  • SBC-RSP stands for SS Basic Capabilities
    Response.
  • Negotiate basic capabilities.
  • The SS informs the BS of its basic capabilities
  • IEEE 802.16
  • Worldwide interoperability of Microwave Access
    High demand for last-mile broadband access.
  • Provide hight-speed internet access to home and
    business subscribers, without wires
  • Uplink Channel
  • Uplink Channel Descriptor (UCD)
  • Configuration Change Count
  • Determine whether the content is changed
  • Mini-Slot Size
  • Units of Physical Slot, Allowable n2m
  • Uplink Channel ID
  • Arbitrarily Chosen by the BS
  • Unique within the MAC-Sublayer domain
  • Uplink Map (UL-MAP)
  • Allocates access to the uplink channel
  • Authorize/key exchange
  • For authorization procedure
  • PKM-REQ /PKM-RSP management message
  • PKM ( Privacy Key Management ) Protocol
  • REG-REQ / REG-RSP management message
  • Initialization registration procedure
  • The process by which the SS is allowed entry into
  • network and becomes manageable

23
Frame structure of IEEE 802.16
24
Step to Network Access
parameter
SS
7. Management msg
parameter
BS
1. SCAN for a downlink channel
2. DL-MAP
3.DCD
4. Adjust local parameter
8. Establish time of day
6.Authorize/key exchange
5. SBC-REQ
7. REG-REQ
25
Step to Channel Access
SS
BS
DATA
Time slot
DATA
26
Outline
  • VOIP Protocols
  • IEEE 802.16 Introduction
  • Voice over Ethernet via IEEE 802.16

27
Voice over Ethernet via IEEE 802.16
  • The new business opportunities created by the
    advance of new technologies -less expensive than
    3G technologies (e.g., Wi-fi, WiMAX).
  • VoIP is considered as a compulsory service (free
    or not-free of charge) in addition to any other
    services.
  • Compared with traditional circuit-switched
    telephone system, many people believe voice over
    packet-switched networks has the potential of
    delivering a more cost-effective service
    solution.

28
Voice over Ethernet via IEEE 802.16
  • Downlink
  • The Base Station (BS) packs the incoming VoIP
    packets for several users into a downlink (DL)
    burst.
  • Transmits burst to the Subscriber Station (SS) in
    a single DL-subframe.
  • The SS extracts its packets and forwards them to
    the VoIP applications in the Ethernet.
  • Uplink
  • upstream VoIP packets into an uplink (UL)
    burst.
  • Transmits them to the BS in a single UL-subframe.
  • The BS then extracts the original VoIP packets
    and forwards them to the Internet

29
Transmission Time of a fixed-size frame
  • T is the transmission time of a fixed-size frame.
  • P is the transmission time of the preambles,
  • B is the transmission time of broadcast messages
  • RC is the register contention time.
  • (V MAC) is the transmission time of a voice
    payload with MAC header
  • C be the maximum number of VoIP sessions that can
    be supported by IEEE 802.16 point to-point
    connection (C 60).

30
VoIP characteristics and requirements
  • Requirements for Voice
  • Typically target a loss rate of 0.25 percent or
    less.
  • One-way latency should be no more than 150 ms.
  • Jitter should be less than 10 ms.
  • Multiple VOIP user multiplexing is critical
  • OFDMA 32 users in every 5 ms per sector
  • HSDPA 4 users in every 2 ms per sector
  • DO-A 8 users in every 1.667 ms per sector
  • VoIP traffic
  • Strict delay latency requirement (RF delay 70 ms
    in simulation)

31
Outline
  • VOIP Protocols
  • IEEE 802.16 Introduction
  • Voice over Ethernet via IEEE 802.16
  • QoS Strategy for VoIP Services over IEEE 802.16

32
QoS Strategy for VoIP Services over IEEE 802.16
  • Qos ( Quality of Service )
  • Associate a packet with a service flow.
  • Service Flowunidirectional flow of packets that
    provides a particular Qos.
  • Support Quality of Service
  • Unsolicited Grant Service - UGS
  • Real-Time Polling Service rtPS (Real-Time
    Polling Service)
  • Non-Real-Time Polling Service nrtPs
  • Best Effort BE
  • UGS -
  • Real-time service flows
  • Periodic, fixed size grants
  • Avoid overhead and latency of frequent SS
    redundant
  • Meet the continuous need of service flow
  • T1 / E1 / VoIP
  • rtPS -
  • Real-time service flows
  • Variable size data
  • MPEG
  • nrtPs -
  • Non real-time service flow
  • Variable size data
  • Best Effort BE
  • Efficient service to best effort

33
QoS functions within the BS and SSs
34
The Effect of factors on Speech Quality
  • Loss
  • Comparative measure of packets transmitted and
    received to the total number of packets that were
    transmitted.
  • Delay
  • The finite amount of time it takes a packet to
    reach the receiving endpoint after being
    transmitted from the sending endpoint.
  • Jitter
  • The difference in the end-to-end delay between
    packets.
  • Throughput
  • The available user bandwidth between an ingress
    point and an egress point
  • Voice transmission over wireless brings along
    with a it a big problem of packet delay or
    latency.

35
Voice Quality computational scheme
  • R-factor
  • The E-model produces a single value
  • Derived from a variety of factors, eg. Delay
  • The range from 0 (extremely poor) to 100 (high
    quality)
  • Any R-factor below 50 is unacceptable
  • Three main variations of R-factor
  • Call quality estimate, RCQE .
  • Listening quality estimate , RLQE .
  • Network performance estimate , RNPE .
  • E-Model -
  • Provides a prediction of the expected voice
    quality
  • Originally the E-Model was intended for use in
    network planning and design.

36
R-factor
  • In the E-Model several different parameters
    affecting the quality of a conversation are taken
    into account.
  • R0 is the basic signal-to-noise ratio
    (environmental and device noises).
  • Is accounts for the impairments on the coded
    voice signal (loud connection and quantization)
  • Id represents the effect of delay
  • Ie the effect of low bit rate codecs
  • A is the advantage factor, corresponding to the
    user allowance due to the convenience in using a
    given technology.

37
Committed Information Rate and Maximum
Information Rate
  • Two main parameters are used in order to support
    service differentiation at the higher layers
  • Committed Information Rate (CIR)
  • The CIR parameter for a WiMAX system is the
    bit-rate that the network agrees to accept from
    the user.
  • Rmax - the maximum traffic rate available at the
    WiMAX Downlink Air Interface.
  • CIRk and MIRk - the request of the k-th SS2.
  • Maximum Information Rate (MIR)
  • regulates the maximum allowed peak rate of a
    connection.

MIR and CIR are specified for each SS according
to the negotiated Service Level Agreement (SLA)
38
VOIP service flow allocation
  • RBE (bit/s) - BE Service Rate.
  • BS can provide to the m-th TCP service flow
  • RTCP(m) be the service rate that the BS can
    provide to the m-th TCP service flow.
  • RRT (bit/s) - Real Time (RT) Service Rate.
  • usual assumption is that the BE flows are TCP
    friendly
  • RVoIP(m) is the service rate that the BS provides
    to the m-th VoIP service flow
  • Ntot - total number of downstream service flows
  • consisting of NVoIP(VoIP flows) and NTCP(TCP
    persistent connection).
  • The opposite case
  • when the aggregate of the CIR requested by VoIP
    sub-scribers exceeds Rmax
  • WiMAX network is deterministically lower than
    Rmax (no congestion occurs).

1. ???????????property? voice service flow
2. ??BS busy???? ,VOIP signaling (BE property?
service)????BS??.
3. BE service?property???, RTCP??RVoIP???????BS??
39
Performance evaluation of WiMAX under VoIP Traffic
  • Deployed in Turin, Italy, within the national
    experimentation on WiMAX coordinated by
    Fondazione Ugo Bordoni (http//wimax.fub.it).
  • The distance between the BS and SS1, SS2 and SS3
    is 8.4 km, 8.5 km and 13.7 km, respectively.
  • The equipment operates using a 3.5 MHz wide
    channel in FDD mode, The SSs work in under FDD
    half-duplex mode. 64 QAM, for each connection.
  • The average signal-to-noise ratio is above 30 dB.
  • All nodes run Linux with a 2.4 kernel.

40
Performance evaluation of WiMAX under VoIP Traffic
  • VoIP codecs feed RTP packet flows and three
    commonly used codecs (G.711,G.729.2 and G.723.1)
  • rtPS services are used for VoIP connections.
  • TCP-controlled traffic is mapped in the BE class.

41
Packet loss rate of VoIP flows per SS using
different codecs
42
Average Delay vs Number of SS VoIP flows
G.729.2 packet generation rate, coupled to the
large overhead of packet headers of the
RTP/UDP/IP/MAC protocol stack ( 45 for the
G.729.2).
43
Average Throughput per VoIP session versus an
increasing number of VoIP per SS
44
Average R-factor versus number of SS VoIP flows
45
Packet Delay pdf
46
Outline
  • VOIP Protocols
  • IEEE 802.16 Introduction
  • Voice over Ethernet via IEEE 802.16
  • QoS Strategy for VoIP Services over IEEE 802.16
  • Conclusions

47
Conclusions
  • We discussed the transmission of VoIP and IMS
    over WiMAX.
  • The working of a Softswitch which makes VoIP over
    WiMAXpossible.
  • WiMAX is to ensure large area coverage and rather
    inexpensive equipment at the subscriber side.
    Modern requirements to wireless connectivity
    include mandatory QoS guarantees for a wide set
    of real-time applications.
  • Wireless Service Providers (WSPs) use WiMAX
    networks to provide connectivity to both
    residential (voice, data and video) and business
    (primarily voice and Internet) customers.

48
Outline
  • VOIP Protocols
  • IEEE 802.16 Introduction
  • Voice over Ethernet via IEEE 802.16
  • QoS Strategy for VoIP Services over IEEE 802.16
  • Conclusions
  • References

49
References
  • WiMAX05 WiMAX Forum "Can WiMAX Address your
    Applications?
  • http//www.wimaxforum.org/news/downloads/Can_WiMAX
    _Address_Your_Applications_final.pdf
  • JDSU03JDSU"White Paper VoIP Overview"-JDSU
    Corporation 2003
  • http//www.jdsu.com/test_and_measurement/technical
    _resources/product_documents/whitepaper/voipterm_w
    p_acc_tm_ae_1205.pdf
  • Performance Evaluation of a WiMAX Testbed under
    VoIP Traffic http//portal.acm.org/citation.cfm?id
    1161009jmpcitcollGUIDEdlGCFID15151515CFT
    OKEN6184618
  • C. Cicconetti, L. Lenzini, E. Mingozzi, and C.
    Eklund.Quality of service support in IEEE 802.16
    networks. IEEE Network Magazine, 20(2)5055,
    March 2006.
  • F. De Pellegrini, D. Miorandi, E. Salvadori and
    N. Scalabrino.QoS Support in WiMAX Networks
    Issues and Experimental Measurements. Technical
    Report 200600009, CREATE-NET, 2006.
  • C. Hoene, H. Karl, and A. Wolisz. A perceptual
    quality model intended for adaptive VoIP
    applications Research articles. Int. J. Commun.
    Syst., 19(3)299316, 2006.

50
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