Access Networks : Voice Over IP

1
Access Networks Voice Over IP

• Miriam Allalouf
• 2006 Semester B

2
Voice Over IP

• Real-Time Interactive Audio and Video
• Internet Phone class of application allows
  people to use audio/video to communicate with ach
  other in real.

3
Voice Over IP

• Voice vs. Data (1)- Tolerance for Delay
• The delay between the sender to the receiver no
  more than around 300 ms.
• The delayed created by The time it takes for As
  speech to get to B The time B hears the speech
  The time B speaks The time it takes the
  response to get back to A.
• If it is above 400 or 500 ms it becomes strange
  and A hears lot of silent (as half duplex).
• Less than 150ms is negligible.
• The data does not care how long it takes the
  trip.
• Normal phone 10 ms/kmile ? 30 ms cost-to-cost
• G.729 10 ms 5 ms look ahead 10 ms
  computation 25 ms processing delay
• G.723.1 100 ms one way delay

4
Voice Over IP

• Voice vs. Data (2)
• Tolerance for Delay and the effects on queues
• Voice requires short queues so the queue length
  will not increase the delay. It can provide an
  overflow with packet loss.
• The data queue are large for bursts handle and to
  prevent packet loss in overflow condition.
• Frame length large MTU can block voice packets
  from higher priority ? Fragmentation

5
Voice Over IP

• More Voice needs Delay Jitter
• Jitter Time variation of packets arrivals at
  the receiver
• Jitter buffer 40-60 ms
• How To overcome Delay Jitter
• Receiver synchronous playout of voice chunks
• Seq number time stamp
• Delaying playout
• Fixed Delay (tradeoff between delay and loss)
• Adaptive Delay

6
Voice Over IP

• Delay Jitter (2)
• How To overcome Delay Jitter at the receiver
  Adaptive Delay Estimation of Network delay and
  variance Ramjee 1994
• ti time stamp of the ith packet generation
• ri the time packet i received by receiver
• pi the time packet i played by receiver
• End-to-end delay of packet I is ri-ti
• di average nw delay calculated
  (1-u)di-1u(ri-ti) for constant u.
• vi average delay deviation (1-u)vi-1u(ri-ti-di)
  for constant u.
• Algorithm for first packet in spurt pi
  tidiKvi ,, qi pi-ti
• For pj in the same spurt pj tjqi
• The receiver can decide if it is a new spurt
  (seq and time stamp)

7
Voice Over IP

• Voice vs. Data (3)Tolerance for Errors and Loss
• The voice exhibits a high tolerance for errors.
  In case of error or loss the further voice
  production is not affected.
• 1-20 be tolerated depends on encoding and
  receivers behavior
• The data traffic has low tolerance for errors.
  One corrupted bit changes the meaning of the
  data.
• UDP is accepted for voice Sequence number, Time
  Stamp. What about TCP?

8
Voice Over IP

• Why Voice Over IP is so important?
• IP is everywhere ? Integration of Voice and Data
  for the WEB server needs, Bandwidth
  Consolidation over the IP layer (in contrast to
  the chanellized voice and data service unit (DSU)
  inefficiency. No bandwidth for the whole
  conversation time.
• The modern analogue-to-digital operations can
  produce a speech channel at about 4.8-8 kb per
  second instead of traditional 64kbps ratio of
  18.
• Tariff arbitrage and beyond Bypass the PSTN
  networks and use instead the internet backbone.
  This approach bypass the high tariffs of the
  Telco's (long distance) to lower costs of
  untariffed internet.

9
Voice Over IP

• Internet (IP world) attributes

10
Voice Over IP

• IP-based telephony improvements
• Better processing technology
• Signal processing
• Switching and transmission.
• PSTN improvements
• Reliability
• Ease of use and coverage
• Number of lines up to 1 billion , immediate dial
  tone and extension of the telephony networks to
  hold all the cellular calls.
• The ability for the PSTN and IP interconnection
  is must!!

11
Voice Over IP

• How to evaluate Packetized voice?
• Bandwidth Requirements
• Number of bit required to pass the voice channel.
  Consists of voice signals , RTP header, UDP
  header, IP header, L7 header, Voice codec header.
• Bandwidth reservation RSVP
• Multiplexing multiple phone calls in a packet
• Voice compression load reduction
• Computational Efforts
• The complexity and expense that are needed for
  the voice support. The number of the instruction
  per second (MIPS).
• The amount of memory.
• More Issues
• Address translation Phone to IP Directory
  servers

12
Voice Over IP

• Different VoIP scenarios

13
Voice Over IP

• Different VoIP scenarios Hybrid systems

14
Voice Over IP

• Voice Services Over IP

15
Voice Over IP

• Hybrid Voice Services
• When 2 different phones are connected via the
  PSTN all the conversions have to be done.
• It is done in the access network.
• It requires mapping of media and media control,
  and signaling channels.
• When the phones are both IP but pass through PSTN
  or both ordinary analog phones but pass via IP
  networks the conversions have to be done in the
  backbone networks.

16
Voice Over IP

• Inter-Working Planes
• The IWF is needed to perform all data or
  signaling adaptation. It is located between the
  NWs or between the terminal and the network
• Voice User Plane Media adaptation
• Voice Control Plane Signaling adaptation

17
Voice Over IP

• Media Adaptation Voice Data
• Voice Data adaptation into the data channels of
  the further NW.
• PSTN media streams as PCM termination ??
• RTP packets generation

18
Voice Over IP

• Media Adaptation

19
Voice Over IP

• Hybrid Systems Signaling and Control
• The IWF gateway should translate the signaling
  from packet-based network to PSTN ISDN and
  vice-versa.
• The Q.931 is used in both IP and ISDN. Still
  due to a specific definition for the ISDN the 931
  peer is created in the ingress of the PSTN.
• Another approach is to provide signaling gateway
  that manage or talk to the media gateways the
  signaling gateway will get ss7 and will talk to
  the media gateway in H.323 or other protocol as
  MGCP Media gateway control protocol.

20
Voice Over IP

• Signaling Adaptation
• Processing and translating of incoming signaling
  messages.
• Call setup and clearing, Address translation

21
Voice Over IP

• Signaling Adaptation - SS7 interoperability
• From historical reasons the gateways are
  connected to the networks as terminals. If the
  gateway wants to be part of the SS7 protocol or
  to receive it appropriately it should be a
  network node (there is an example where two call
  centers (IP networks) connected virtually and
  there is a PSTN between them via 2 PSTN/IP
  gateways. If one is busy and the gateway wants to
  transfer the calls automatically to the other
  call center it should have an access to the SS7
  signaling otherwise it is a big mess.
• The SS7 is center to the operation of the PSTN
  and hence it is very important to the the Telcos
  to provide the ss7 to the gateways providers.

22
Voice Over IP

• Signaling Adaptation

23
Voice Over IP
24
Voice Over IP

• H.323 System components
• Each H.323 system contains few Zones
  interconnected via the WAN. Each Zone contains
  few LANs.
• H.323 Terminal The terminal is an endpoint
  entity.Provides the following
• Real-time two-way communication with other H.323
  TEs, GW or MCU.
• Audio
• Moving color Video pictures
• Data
• Control and Indication
• A call to another TE may be set via the GK or not.

25
Voice Over IP

• H.323 System components
• GateKeeper Optional component that provides
  call control services to the terminals within a
  specific Zone. It provides to the TE, GW and MCU
• Addressing resolution
• Authorization,Access control
• Admission control bandwidth management
• GWs location
• Gateway GW is an endpoint entity (few per
  zone). Provides the following
• Real-time two-way translation of formats and
  control procedures between H.323 TEs on the
  packet-based NW and PSTN terminals.

26
Voice Over IP

• H.323 System components
• MCU Multipoint control unit
• The MCU is an endpoint that provides the
  capabilities for three and more terminals and GWs
  to participate in a multipoint conference.

27
Voice Over IP

• H.323 System Protocols for signaling
• RAS (Registration,admission and Status)
  protocol (H.225.0) is used between the H.323
  endpoint (TE or GW) and gatekeeper
  communication.
• Discovering the GateKeeper
• Registration in the gatekeeper (for updating its
  tables)
• Admission control an BW allocation for this
  specific call.
• Call clearing
• GateKeeper inquires endpoint status via RAS
• GateKeepers communication for address resolution
  across zones

28
Voice Over IP

• H.323 System Protocols for data and signaling
• Protocol Q.931 Signaling protocol between two
  H.323 terminals It is a variation of Q.931
  protocol defined for PSTN thus to ease the
  interworking with PSTN/ISDN
• Call setup announcement to the destination for
  the ends settings
• Protocol H.245 Signaling protocol between two
  H.323 terminals for media negotiation
  capabilities as audio/video codecs
• Media Resources reservation
• Determines master-slave relationships between
  the endpoints
• logical channel assignment

29
Voice Over IP

• H.323 System Protocols for data
• The voice data stable call
• RTP (Real time transmission protocol) over UDP
  voice packets
• Associated with RTCP Real time control protocol

30
Voice Over IP

• H.323 Voice Terminal

31
Voice Over IP

• Signaling Adaptation

32
Voice Over IP

• H.323 Control Protocols
• Direct routed call
• GK-routed call Good for IP-telephony can be
  controlled and charged
• With GK it takes 7 phases to setup and teardown a
  call
• H.323V1 (5/96)
• 7 or 8 round trips
• H.323V2 (2/98) -
• 2 round trips
• Combined Q.931 and H.245 phases

33
Voice Over IP

• H.450 Standard for Telephony services
• H.450 This standard is used in H.323system
• Services Call transfer,call forwarding,call
  deflection,call hold, call park and pickup
  message waiting indication, name indication, call
  completion
• Relies on the end-to-end signaling TEs execute
  most of the service logic. In contrast to the
  PSTN where most of intelligence was in the
  exchanges.
• Service deployment without H.450
• H.450 addition required TE upgrade
• Intelligent NW (IN) is possible by providing the
  GK all H.450 capabilities and using GK-routed
  call.

34
Voice Over IP

• H.323 System GW limitations
• H.323 Gateway provides all functionality in one
  box
• Signaling conversion,Call control,Media
  translation
• limitations
• Scalability-H.323 GW can support maximum of tens
  thousand users because of its intensive media
  processing echo cancellation,silence compression
• no SS7 connectivity (inhibits IWF w PSTN)
• Availability H.323 GW concentrates all call no
  fail over.
• Two stage dialing GW number dialing callee
  number dialing

35
Voice Over IP

• Solution H.323 GW functionality decomposition
• Separation of signaling and media processing
  functions ETSI TIPHON
• Signaling Gateway (SG) - for signaling
  conversion IP/PSTN
• Media Gateway (MG) - Media transformation PCM
  ?? RTP, Voice compression,Echo cancellation,Silenc
  e suppression,encryption, Services as voice
  recognition and paying announcements.The MG can
  be located in the residential site towards the
  local loop or before the PBX as the access or in
  the backbone between IP and PSTN NW as trunking
  media GW.
• Media Gateway controller (MGC) for Call control
  located between the MG,SG and GK.
• Call handling (state maintenance) for MG
• Gets PSTN signaling from SG and IP signaling from
  GK or from IP TE (Q.931,H.245)
• Manages NW level resources availability (IP and
  trunks capacity, routing decision)

36
Voice Over IP

• Protocols
• MGCP Control protocol between the MGC and an
  MG.
• Common standard of ITU and IETF called Megaco
  H.248
• Did the decomposition solve the problem?
• Scalability- The MGC handles few MG boxes and
  provides virtually more lines.
• SS7 connectivity provides One stage dialing
• Availability Multiple MGCs can handle the MG
  fault tolerance.

37
Voice Over IP

• MGCP and H.323
• The Gatekeepers will use the MGCP to control
  large MGs
• Still intelligence is kept in the endpoints
  (H.323)
• MGCP instead H.323 signaling
• Preferred when a centralized approach is needed
  thin clients inexpensive CPE (no need for new
  services upgrade)

38
Voice Over IP

• SIP
• SIP (Session Initiation protocol) The SIP
  server provides name-to-address resolution and
  user location
• Upon call setup, the MGC (Call agent) sends a
  request to the SIP server with the call name and
  gets back the IP address for routing. It can send
  few messages and get different IP addresses and
  then decide on the preferred one.
• Supports User location,call setup call transfer
• Works with other IP protocols RSVP,RTP/RTCP/RTSP,
  SAP,SDP
• SIP advantages Stateless, HTTP
  compatible,Opaque body message, Uniform resource
  identifier (WEB connectivity).
• SIP and H.323
• Co-exist or replace

39
Voice Over IP

• What is missing?
• IP QoS

40
Access Networks Telephony NW

• Abbreviations
• VOIP-Voice over IP
• PCM Pulse Code Modulation
• SIP-Session Initiate Protocol
• RTP-Real time protocol
• RTCP-Real time control protocol
• RTSP-Real time streaming protocol
• MG-Media Gateway
• MGC -Media Gateway Control
• MGCP- -Media Gateway Control Protocol

41
Access Networks Introduction

• References
• Voice Over IP, Uyless Black
• www.von.com
• www.iptelephony.org
• www.ietf.org