Holly Voice Platform Overview - PowerPoint PPT Presentation

Loading...

PPT – Holly Voice Platform Overview PowerPoint presentation | free to view - id: 228b7-ZDQ1N



Loading


The Adobe Flash plugin is needed to view this content

Get the plugin now

View by Category
About This Presentation
Title:

Holly Voice Platform Overview

Description:

SpeechTEK 2006 Voice Over IP Tutorial. Andrew Hunt, Ph.D. VP Engineering, Holly Connects ... hunt_at_holly-connects.com. Web: http://www.holly-connects.com ... – PowerPoint PPT presentation

Number of Views:180
Avg rating:3.0/5.0
Slides: 128
Provided by: andre92
Category:

less

Write a Comment
User Comments (0)
Transcript and Presenter's Notes

Title: Holly Voice Platform Overview


1
SpeechTEK 2006 Voice Over IP Tutorial
Andrew Hunt, Ph.D. VP Engineering, Holly Connects
2
Welcome!
Who are you?
3
(No Transcript)
4
Timing
  • 830am Start
  • 1000-1030am Coffee break
  • 1200-100pm Lunch
  • 230-300pm Coffee break
  • 430pm Close

5
Agenda
  1. Welcome Introductions
  2. Why Voice Over IP?
  3. Brief History of Telephony
  4. Digital Voice
  5. Voice Over IP Basics
  6. VoIP Protocols
  1. SIP Session Initiation Protocol
  2. RTP Real-time Transport Protocol
  3. Network Issues and Design
  4. VoIP and Speech Recognition
  5. VoIP and Mobile Telephony
  6. Closing

6
Objectives
  • Informative
  • Relevant
  • Interesting
  • Interactive

7
Why Voice Over IP?
8
Module Overview
  • Why Voice Over IP?
  • What is it?
  • Cost
  • Functionality flexibility
  • Mobility

9
VoIP Definition
  • Definitions
  • Voice Over IP is the use of the internet,
    intranets and other IP networks for the delivery
    of voice conversations
  • Internet Protocol (IP) is a protocol used for
    communicating data across a packet-switched
    network (specifically IPv4 or IPv6)
  • Many VoIP protocols exist focus today on SIP and
    related protocols for use in speech recognition
    and IVR contexts

10
Why VoIP?
  • Cost
  • Global data traffic exceeded voice traffic in
    late 1990s
  • Telco charges revenue largely from voice
    traffic
  • Migration to shared networks
  • Single network for voice and data
  • Utilize spare capacity in many data networks
  • Inwards-out approach to migration
  • Traditional voice carriage costs to business
    being driven down by VoIP
  • Residential advantages
  • Free services Skype etc.
  • Lower cost services Vonage, Skype etc.
  • Regulatory and service capability issues are
    evolving

11
Why VoIP?
  • Functionality and Flexibility
  • Virtualization mobility move calls and agents
    locally, nationally, globally
  • Match traditional telephony functions
  • Transfers, voice mail, conferencing, redial,
    speed-dial, forwarding etc.
  • Integration with software and internet services
  • Availability notification (IM)
  • Instant messaging
  • Multi-media services video, data files etc.
  • Options for traditional and VoIP telephony to
    co-exist and migrate steadily

12
Why VoIP?
  • Mobility
  • Make VoIP calls virtually anywhere in the world
  • Number portability landline, mobile, internet
  • Synchronization of address books and contacts

13
Brief History of Telephony
14
Module Overview
  • Brief History of Telephony
  • Telephony switching
  • Basic concepts of telephony
  • Call establishment
  • Circuits
  • Switching
  • Migrating from analogue to digital

15
Telephony Switching and Circuits
Alexander Graham Bell
Thomas Watson
16
Telephony Switching and Circuits
  1. Caller picks up
  2. Caller dials
  3. Callee phone rings
  4. Callee answers
  5. Circuit established
  6. Conversation
  7. Hang-up tear-down

17
Telephony Switching and Circuits
Exchange
Exchange
Exchange
Exchange
Exchange
18
Telephony Switching and Circuits
PSTN Public Switched Telephony Network
  1. A-party picks up
  2. A-party dials
  3. B-party rings
  4. B-party answers
  5. Circuit established
  6. Conversation
  7. Hang-up tear-down

19
Telephony Switching and Circuits
  • Pre-Digital Analogue Era
  • 1878 New Haven, Connecticut
  • Worlds first commercial telephone exchange
  • Built from carriage bolts, handles from teapot
    lids and bustle wire
  • Cost 40 including the furniture
  • 1891 Topeka, Kansas
  • Almon Strowger, an undertaker, patented the
    Strowger switch
  • Automation of the telephone circuit switching by
    decadic pulses
  • 1950 onwards Crossbar switches
  • 1964 Dual Tone Multi-Frequency (DTMF)
    introduced

20
Telephony Switching and Circuits
  • Pre-Digital Analogue Era
  • Infrastructure
  • Local wiring to each phone
  • System of local, regional, national and
    international exchanges
  • Shared connections between exchanges
  • Call establishment
  • Numbering scheme (many iterations)
  • Voice ? decadic pulsing ? DTMF
  • Mapping of numbering scheme to the exchanges
    (e.g. CAstle22)
  • Call communication
  • Dedicated circuit per call

21
Telephony Switching and Circuits
  • Digital Era
  • Digital started with the core telco networks
  • Efficiency on long-distance carriage
  • Efficiency of solid state switching technology
  • Migrated to local exchanges
  • What is digital voice?

22
Digital Voice
23
Module Overview
  • Digital Speech
  • Sampling creating digital audio
  • CODECs Compression and companding
  • Sharing channels Time Division Multiplexing
    (TDM)
  • Standard digital data links E1 T1

24
Telephony Packet Switch Network
0101100
0101100
0011100
0011100
1100110
1100110
1101100
1101100
0001100
0001100
PSTN Public Switched Telephony Network
0101100
0101100
0011100
0011100
1100110
1100110
1101100
1101100
0001100
0001100
25
Sampling Theory
26
Sampling Theory
27
Sampling Theory
28
Sampling Theory
29
Sampling Theory
Time (msec) 0 Value 0
1 29
2 57
3 82
4 102
5 117
6 125
7 126
8 120
9 108
10 89
11 66
12 39
13 9
14 -20
15 -49
16 -75
17 -97
18 -114
19 -124
20 -127

30
Sampling Theory Companding
Non-linear sampling ? better noise/error
G.711 µ-law for North America and JapanG.711
A-law for Europe and rest of the world
31
Sampling Theory
Measure G.711 Telephony Compact Disc
Sampling rate 8,000 Hz 44,100 Hz
Frequency range Low 3.5kHz (4kHz max) Lower 22 kHz
Sample type 8 bit A-law / µ-lawMono 16-bit linear PCMStereo
Signal-to-noise ratio lt 70dB 96 dB
Data bandwidth 8 kByte/sec 176 kByte/sec
Perceived quality Telephony Great
32
Digital Transport
  • 102
  • -18
  • -48
  • -60
  • -12
  • 10
  • 52
  • 87
  • 92
  • 85
  • 49
  • 10

TDM Time Division Multiplex
  • 102
  • 102
  • -18
  • -48
  • -49
  • -18
  • -48
  • -60
  • -12
  • -60
  • -12
  • 85
  • 49
  • 10
  • 10
  • 10
  • 52
  • 87
  • 92
  • 10
  • 52
  • 87
  • 92
  • 85
  • 102
  • -18
  • -48
  • -60
  • -12
  • 85
  • 49
  • 10
  • 10
  • 52
  • 87
  • 92

Packet
Latency
33
Digital Transport
  • E1
  • World (ex. NA and Japan)
  • 2.048 Mbit/s full duplex
  • 32 time slots 32 channels of 8-bit x 8 kHz
  • 1 time slot reserved for framing
  • 1 time slot is typically reserved for signalling
  • 30 time slots for voice communications
  • E3 16 x E1 480 channels
  • T1
  • North America and Japan
  • 1.536 MBit/s full duplex
  • 24 time slots 24 channels of 8-bit x 8kHz

34
Compression
  • Compression of voice and audio
  • CODEC COmpression - DECompression
  • Reduce bandwidth for the audio signal more
    channels on the same transport
  • Lossless vs. lossy algorithms
  • Latency impact
  • CPU impact
  • Quality impact
  • Speech recognition impact

35
Sampling Theory
CODEC Characteristics Description
ITU-T G.711 Sample rate 8kHzSample size 8-bit A-law/µ-law Bandwidth 64kbit/s Standard telephony quality with companding
ITU-T G.726 Sample rate 8kHzSample size 2, 3, 4, 5-bitBandwidth 16, 24, 32, 40 kbit/s Adaptive Delta PCM (ADPCM). Supercedes G.721 G.723
ITU-T G.728 Bandwidth 16 kbit/sDelay 5 samples, 0.625 ms LDCELP Low Delay Code Excited Linear Prediction
ITU-T G.729 Bandwidth 8 or 6.4, 11.8 kbit/sDelay 10 ms chunks CS-ACELP Conjugate-Structure Algebraic-Code-Excited Linear Prediction
ITU-T G.722 Sample rate 16kHzSample size 14-bit Bandwidth 32-64kbit/s Standard wideband speech ADPCM codec.Non-telephony CODEC
36
Digital Transport
  • CODECs can increase channel capacity
  • 1 E1/T1 channel 64 kbit/s
  • 1 channel of G.711
  • 2 channels of G.726 _at_ 32 kbit/s
  • 8 channels of G.729 _at_ 8 kbit/s
  • 1 T1 24 channels
  • 24 channels of G.711
  • 48 channels of G.726 _at_ 32 kbit/s
  • 196 channels of G.729 _at_ 8 kbit/s
  • BUT, CODECs can reduce voice quality

37
Digital Transport
  • Advanced topics
  • Unreliable tone transmission on some CODECs
  • DTMF, Fax, Modem etc.
  • Use out-of-band communication (more later)
  • Silence suppression
  • Voice Activity Detection
  • Replace by simulated background noise
  • e.g. G.729 Annex B
  • Comfort noise generator (CNG)
  • Played when a communication channel fails
    temporarily
  • Usability Reduces hang-up on temporary outages

38
(No Transcript)
39
Voice Over IP Basics
40
Module Overview
  • Voice Over IP Basics
  • Internet and intranet for voice communication
  • Challenges of the internet protocols
  • Quick guide to the internet protocols TCP/IP,
    UDP
  • Applying internet protocols to session management
    voice carriage

41
Telephony Packet Switch Network
0101100
0101100
0011100
0011100
1100110
1100110
1101100
1101100
0001100
0001100
PSTN Public Switched Telephony Network
0101100
0101100
0011100
0011100
1100110
1100110
1101100
1101100
0001100
0001100
42
Voice Over Internet Protocol
0101100
0101100
0011100
0011100
1100110
1100110
1101100
1101100
Intranet OR Internet
0001100
0001100
0101100
0101100
0011100
0011100
1100110
1100110
1101100
1101100
0001100
0001100
43
Challenges
  • Call establishment protocols
  • Connect A-party and B-party
  • Perform advanced telephony functions
  • Audio transport protocols
  • Get the packets from A-to-B / B-to-A on time
  • Latency
  • Packet loss
  • Jitter
  • Quality of service

44
The Problem
  • The internet was not designed to carry real-time
    voice traffic!!! (well almost)

45
Internet Protocol Suite Stack
Application
DNS, FTP, HTTP, SMTP, SNMP, TELNET, SIP, RTP,
H.323
Layer 4
Transport
Layer 3
TCP, UDP, SCTP, DCCP, IL, RUDP,
Network
Layer 2
IP (IPv4, IPv6)
Link
Layer 1
Ethernet, Wi-Fi, ATM, Frame Relay
46
Voice Over Internet Protocol
Application
Internet
Application
47
Internet Protocol Suite Stack
Application
Application
Peer-to-peer connection
Transport
Transport
Network
Network
Network
Network
Link
Link
Link
Link
48
Internet Protocol Suite Stack
Application
Application
Peer-to-peer connection
0101100
Transport
Transport
Network
Network
Network
Network
0101100
0101100
Link
Link
Link
Link
49
Internet Protocol Suite Stack
Application
Application
Peer-to-peer connection
0101100
Transport
Transport
Packet Loss
Network
Network
Network
Network
0101100
0101100
Link
Link
Link
Link
50
Internet Protocol Suite Stack
  • TCP/IP Transmission Control Protocol
  • Transport protocol
  • One of the core protocols of the Internet
    protocol suite 75 of all traffic
  • Applications on networked hosts can create
    connections to one another using TCP for exchange
    of data or packets.
  • Guarantees reliable and in-order delivery of data
    from sender to receiver
  • Distinguishes data for multiple, concurrent
    applications on the same host
  • TCP supports many of the most popular application
    protocols including HTTP (Web), Email and SIP
  • Send a stream of bytes through a virtual pipe
  • Utilizes sequence numbers, acknowledgement,
    timeout, retransmission

51
Internet Protocol Suite Stack
  • TCP/IP for Voice Over IP
  • Good for session management
  • H.323 / ASN.1 built on TCP/IP
  • Sub-optimal for near-realtime audio transport
  • Latency
  • Jitter
  • Aside Utilized for Skype

52
Internet Protocol Suite Stack
  • UDP User Datagram Protocol
  • Transport protocol
  • One of the core protocols of the Internet
    protocol suite 20 of all traffic
  • Does not provide the reliability and ordering
    guarantees of TCP
  • Datagrams may arrive out of order or be dropped
    by the network
  • Datagram transmission is stateless in the network
  • Lower overhead faster, more efficient, suited
    to time-sensitive comms
  • UDP supports many application protocols including
    DNS and RTP

53
Internet Protocol Suite Stack
  • UDP for Voice Over IP
  • OK for session management
  • End-parties need reliable communication about
    session status
  • SIP utilises UDP with retry behaviour to
    withstand packet loss
  • UDP offers faster setup time than TCP/IP
  • Suited to near-realtime audio transport
  • Utilized by RTP
  • Better latency than TCP/IP (though not ideal)
  • Better jitter than TCP/IP (though not ideal)
  • Packet loss causes poorer audio transmission than
    TCP/IP

54
Internet Protocols for Voice Over IP
  • Internet vs. PSTN
  • Internet has smart terminals and dumb network
  • PSTN has dumb terminals and smart network
  • PSTN dedicates virtual connections for audio and
    session
  • Internet normally creates connections on an
    as-needed basis
  • Internet protocols emerging for traffic shaping
    suited to telephony
  • Network tools also emerging for banishing and
    punishing telephony

55
VoIP Protocols
56
Module Overview
  • VoIP Protocols
  • Overview of the landscape of VoIP protocols
  • Major IETF standards SIP, RTP, RTCP
  • Major ITU-T standards H.323 family

57
VoIP Protocols IETF
Protocol Protocol Description
SIP Session Initiation Protocol Session management on UDP
RTPSRTP Real-time Transport ProtocolSecure RTP Audio/video media delivery on UDP
RTCPSRTCP Real-time Transport Control ProtocolSecure RTCP Out-of-band control protocol for RTP
58
VoIP Protocols
  • About SIP
  • IETF Protocol
  • Standard for initiating, modifying, and
    terminating a user session that may involves
    media elements such as voice, video, instant
    messaging etc.
  • Used widely in telephony environments
  • Supported by numerous IVR platforms
  • Accepted in 2000 as the signalling protocol of
    the IMS architecture
  • Other uses
  • MRCP v2 Media Resource Control Protocol for
    Speech Recognition and Text-to-Speech
  • Microsoft Messenger

59
VoIP Protocols ITU-T
Protocol Description
H.323 Umbrella recommendation for audio-visual comms on any packet networkReferences the following specifications
H.225.0 Protocol to describe call signaling, the media (audio and video), the stream packetization, media stream synchronization and control message formats
H.245 Control protocol for multimedia communication with messages and procedures used for opening and closing logical channels for audio, video and data, capability exchange, control and indications
H.235 Describes security in H.323
H.329 Describes dual stream use in videoconferencing, usually one for live video, the other for presentation
60
VoIP Protocols
  • About H.323
  • Based on ISDN Q.931
  • Suited to internetworking between IP and ISDN /
    QSIG
  • Similar call model to ISDN
  • Used widely in telephony environments
  • Telecommunications backbones
  • Other uses
  • Microsoft NetMeeting

61
Todays Focus
  • Well focus on SIP today

62
SIP Session Initiation Protocol
63
Module Overview
  • Session Initiation Protocol
  • Components of the SIP architecture
  • SIP Messaging
  • Standard SIP exchanges
  • Ring, hold, answer, transfer, consultative
    transfer, conferencing
  • SIP addresses
  • SIP working with RTP for Voice

64
SIP Overview
  • Session Initiation Protocol (SIP)
  • There are many applications of the Internet that
    require the creation and management of a session,
    where a session is considered an exchange of data
    between an association of participants. The
    implementation of these applications is
    complicated by the practices of participants
    users may move between endpoints, they may be
    addressable by multiple names, and they may
    communicate in several different media -
    sometimes simultaneously. Numerous protocols have
    been authored that carry various forms of
    real-time multimedia session data such as voice,
    video, or text messages. The Session Initiation
    Protocol (SIP) works in concert with these
    protocols by enabling Internet endpoints (called
    user agents) to discover one another and to agree
    on a characterization of a session they would
    like to share. For locating prospective session
    participants, and for other functions, SIP
    enables the creation of an infrastructure of
    network hosts (called proxy servers) to which
    user agents can send registrations, invitations
    to sessions, and other requests. SIP is an agile,
    general-purpose tool for creating, modifying, and
    terminating sessions that works independently of
    underlying transport protocols and without
    dependency on the type of session that is being
    established.

65
SIP Overview
  • Session Initiation Protocol (SIP)
  • http//www.ietf.org/rfc/rfc3261.txt
  • Protocol developed by the IETF MMUSIC Working
    Group (now SIP)
  • Scope initiate, modify and terminate an
    interactive user session that involves voice and
    multimedia elements such as video, instant
    messaging and games.
  • SIP 2.0 published as RFC 3261 in 2002
  • Initial release of SIP 1.0 as RFC 2543 in 1996
    (now obsolete)
  • SIP enables device-to-device communication with
    media communication via other protocols
  • SDP Session Description Protocol - RFC 2327
    (describe media capabilities)
  • RTP Real-time Transport Protocol - RFC 3550
    (transport audio, video, media)
  • RTCP Real-time Transport Control Protocol - RFC
    3550 (control transport of media)
  • Standard protocols with high level of product
    interoperability

66
SIP Overview
  • Session Initiation Protocol (SIP)
  • SIP is an application-layer control protocol in
    Internet stack
  • SIP is an device-to-device, client-server session
    signalling protocol
  • SIP establishes sessions for voice and other
    media
  • Allows integration with others services web,
    email, IM
  • Allows presence and mobility services

67
SIP Overview
  • Applications of SIP
  • SIP can convey arbitrary payload
  • Session description
  • Instant messages
  • Pictures (e.g. picture of the caller)
  • Speech recognition control
  • Web pages

68
SIP Overview Devices
Cisco
BlackBerry
Express Talk
Siemens
xTen
Avaya
69
SIP Overview
  • Network Devices
  • SIP Proxy Server
  • Intermediary to relay call signalling
  • SIP Redirect Server
  • Redirects callers to other servers
  • SIP Registrar
  • Accept registration requests from users
  • Maintains users whereabouts
  • SIP IVR
  • SIP PBX

70
SIP Communications
  • SIP Jargon
  • User Agent Client Initiates a communication
  • User Agent Server Respondent to a communication
  • Note device can be both client and server in a
    single session
  • Examples
  • Desktop phone is both a client (makes calls) and
    server (receives calls)

71
SIP Communications
  • SIP Addresses
  • SIP address can make you globally reachable
  • Callees bind to this address using SIP REGISTER
    method
  • Callers use this address to establish real-time
    communication with callees
  • SIP address is a URI address format
  • sipandrew.hunt_at_holly-connects.com
  • sip0282078218_at_asterisk.holly-connects.com
  • sipvoicemail_at_holly-connects.com?subjectwassup
  • Can embed in web pages or place on your business
    card
  • Highlighted text is the public identifier
  • SIP URI address contents
  • Must include host
  • May include user name
  • May include the port number
  • May include others parameters (e.g., transport)

72
SIP Communications
  • Protocol design
  • Similar protocol to HTTP and SMTP
  • Transmission via UDP messages
  • Human-readable messages
  • Simple interaction mechanism
  • User Agent A sends a Control Message to User
    Agent B
  • User Agent B sends a Response Code to User
    Agent A
  • Retry in the event of communication failure

73
SIP Communications
  • SIP Methods (Control Messages)
  • INVITE invite a user agent to a session
  • ACK acknowledge a communication
  • OPTIONS query servers about their capabilities
  • REGISTER register with a SIP Registrar
  • BYE terminate a session
  • CANCEL cancel a session

74
SIP Communications
  • SIP Response Codes
  • 1xx Provisional -- request received, continuing
    to process the request
  • 2xx Success -- the action was successfully
    received, understood, and accepted
  • 200 OK
  • 3xx Redirection -- further action required to
    complete the request
  • 4xx Client Error -- the request contains bad
    syntax or cannot be fulfilled
  • 5xx Server Error -- the server failed to fulfil
    an apparently valid request
  • 6xx Global Failure -- the request cannot be
    fulfilled at any server

75
SIP Communications
  • SIP Extensions (selection amongst many)
  • INFO carry session-related control information
  • RFC 2976
  • e.g. ISUP and ISDN signalling messages
  • REFER refer the recipient to a new resource
  • RFC 3515
  • e.g. Call transfer

76
SIP Simple Peer-to-Peer Session
B
A
  • Simple peer-to-peer SIP session
  • Assumptions
  • A knows address of B
  • A and B can see each other on the network
  • Audio communication
  • Humans are using A and B

77
SIP Simple Peer-to-Peer Session
A
B
User makes a call to B ?
? Play ring-tone to user B
Play ring-tone to user A?
- - - Waiting for answer - - -
? B User answers call
A hears B ?
? B hears A
- - - Call Established - - -
? B hangs up
? B terminates RTP audio
A terminates audio ?
- - - Session Over - - -
78
SIP Simple Peer-to-Peer Session
SIP MessagesStatus codesRTP
A
B
User makes a call to B ?
? Play ring-tone to user B
Play ring-tone to user A?
- - - Waiting for answer - - -
? B User answers call
A hears B ?
? B hears A
- - - Call Established - - -
? B hangs up
? B terminates RTP audio
A terminates audio ?
- - - Session Over - - -
79
SIP Communications
  • SIP Message Example
  • INVITE sip0282078101_at_10.0.0.1135077 SIP/2.0
  • Via SIP/2.0/UDP 10.0.0.35060
  • From ltsip0282078100_at_10.0.0.3gttag40A0C340-2BC
  • To ltsip0282078101_at_10.0.0.113gt
  • Date Fri, 07 Jul 2006 015855 GMT
  • Call-ID FB5F1FE3-C9211DB-B22481EA-DE5FE23A_at_10.0.0
    .3
  • Supported timer,100rel
  • Min-SE 1800
  • Cisco-Guid 4217155242-210899419-2988540394-373082
    5786
  • User-Agent Cisco-SIPGateway/IOS-12.x
  • Allow INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK,
    COMET, REFER, SUBSCRIBE, NOTIFY, INFO
  • CSeq 101 INVITE
  • Max-Forwards 6
  • Remote-Party-ID ltsip0282078100_at_10.0.0.3gtpartyc
    allingscreenyesprivacyoff
  • Timestamp 1152237535
  • Contact ltsip0282078100_at_10.0.0.35060gt
  • Expires 180

Header
To From
Unique call ID
User agent details
Transmission info
Multi-part media content
SDP removed
80
SIP Communications
  • SDP Session Description Protocol
  • RFC 2327
  • Describe media capability of a SIP user agent

Agent Description
v0 oHolly-HVG-4-2 2890844526 2890842809 IN IP4
10.0.0.113 ssip call from the hvg cIN IP4
10.0.0.113 t0 0 maudio 11946 RTP/AVP 8 101 cIN
IP4 10.0.0.113 artpmap0 PCMU/8000 artpmap8
PCMA/8000 artpmap101 telephone-event/8000 afmtp
101 0-16
Agent Address
Audio Capability 1RTP Protocol
G.711 u-law A-law8000Hz
DTMF events 0123456789ABCD
81
SIP Communications
  • SDP Protocol Structure
  • v (protocol version)
  • o (owner/creator and session identifier).
  • s (session name)
  • i (session information)
  • u (URI of description)
  • e (email address)
  • p (phone number)
  • c (connection information)
  • b (bandwidth information)
  • One or more time descriptions
  • z (time zone adjustments)
  • k (encryption key)
  • a (zero or more session attribute lines)
  • Zero or more media descriptions
  • m (media name and transport address)
  • i (media title)
  • c (connection information - optional if
    included at session-level)
  • b (bandwidth information)
  • k (encryption key)
  • a (zero or more media attribute lines)
  • Time description
  • t (time the session is active)
  • r (zero or more repeat times)
  • optional

82
SIP Registration
Registration Process
A
SIPRegistrar
REGISTER
200 OK
- - - Repeat regularly (5, 60 min) - - -
REGISTER Via IPHost From Andrew Hunt
ltsip0282078218_at_ahunt.holly-connects.comgt To
ltsipasterisk.holly-connects.com5060gt CallID
ltltsomething uniquegtgt Expires 3600
83
SIP SIP Session Via a Proxy
B
A
SIP Proxy
User makes a call to B ?
? Play ring-tone to user B
- - - Waiting for answer - - -
Play ring-tone to user A?
? B User answers call
A hears B ?
? B hears A
- - - Call Established - - -
A hangs up ?
A terminates RTP audio ?
? B terminates RTP audio
- - - Session Over - - -
84
SIP SIP Proxy Chaining
A
B
85
SIP Proxy
  • SIP Proxy Function
  • Serve as rendezvous point at which callees are
    reachable
  • Perform routing function
  • Select the next hop or hops when chaining
  • Forking try multiple destinations in parallel or
    sequence
  • Avoid loops when chaining
  • Available capabilities
  • Programmable routing decisions tables
  • Least-cost routing
  • Firewall traversal
  • Direct certain calls to PSTN via gateway (e.g.
    911, local calls)

86
SIP Session Via a PBX
A
B
SIP PBX(e.g. Asterisk)
User makes a call to B ?
100 TRYING
? Play ring-tone to user B
- - - Waiting for answer - - -
Play ring-tone to user A?
? B User answers call
A hears B ?
- - - Call Established - - -
? B hangs up
? B terminates RTP audio
A terminates RTP audio ?
- - - Session Over - - -
87
SIP Session Via a PBX with Redirect (Direct
Audio Link)
A
B
SIP PBX(e.g. Asterisk)
- - - Call Established - - -
- - - Call Continues - - -
? B hangs up
? B terminates RTP audio
A terminates RTP audio ?
- - - Session Over - - -
88
SIP-TDM Gateway
  • Translate signalling messages
  • To/From traditional telephony and VoIP
  • e.g. ISUP ?? SIP/RTP
  • Support heterogeneous environments
  • Staged migration to VoIP
  • Numerous gateway products available

TDM/ISDN
SIP/RTP
PSTN
IP
VoIPGateway
89
SIP-TDM Gateway
PSTN
IP
Signalling gateway
ISDN
Media Gateway Controller
SIP
Media Gateway
RTP
TDM
90
SIP-TDM Gateway
TDM-VoIPBridge
SIP UA
User makes a call ?
? Phone rings
Play ring-tone ?
? Pick-up
- - - Call in Progress - - -
Hang up ?
Terminates RTP audio ?
? Terminates RTP audio
- - - Session Over - - -
91
SIP Sending Auxiliary Information
A
User makes a call ?
INVITE sip0282078101_at_10.0.0.1135077
SIP/2.0 ltltSIP header goes heregtgt Content-type
multipart/mixed boundary"gc0p4Jq0M2Yt08jU534c0p"
MIME-version 1.0 This is a multi-part message
in MIME format. --gc0p4Jq0M2Yt08jU534c0p Content-T
ype application/sdp Content-Length 235 ltltSDP
header goes heregtgt--gc0p4Jq0M2Yt08jU534c0p Conten
t-type image/jpeg PGh0bWwCiAgPGhlYWQCiAgPC9o
ZWFkPgogIDxib2R5PgogI CAgPHAVGhpcyBpcyB0aGUgYm9ke
SBvZiB0aGUgbWVzc2Fn ZS48L3ACiAgPC9ib2R5Pgo8L2h0bW
wCg --gc0p4Jq0M2Yt08jU534c0p
92
SIP Failed Establishment
A
B
?
User makes a call to B ?
?
Timeout ?
- - - Retries - - -
?
Timeout ?
Timeout ?
B gives up ?
93
RTP Real-time Transport Protocol
94
Module Overview
  • RTP Real-time Transport Protocol
  • Really real-time?
  • RTP session overview
  • RTP packets
  • RTP and network transmission

95
Real-time Transport Protocol
  • Standard for real-time transport over IP networks
  • Streaming audio and video
  • Utilised in SIP/RTP and H.323
  • Adopted by 3GPP for next generation cellular
    telephony
  • Widespread use in streaming QuickTime, Real,
    Microsoft
  • RTP assumes
  • Network is dumb and imperfect, end-points are
    smart
  • Network may exhibit delays, jitter, packet loss
    etc.
  • Real-time Transport Protocol is NOT REAL-TIME
  • No end-to-end protocol, including RTP, can ensure
    in-time delivery. This would require the support
    of lower layers (switches, routers etc.)
  • RTP provides functionality suited for carrying
    real-time content, e.g., a timestamp and control
    mechanisms for synchronizing different streams
    with timing properties

96
Real-time Transport Protocol
  • One RTP session transmits one media type
  • Audio / voice
  • Video
  • Multi-media requires multiple RTP sessions
  • RTP session
  • Implements a particular RTP profile
  • Includes an RTP data flow
  • Transports a single media type according to one
    or more payload formats
  • e.g. audio in G.711 format
  • Includes an RTP control protocol flow
  • Providing reception quality feedback, user
    information, etc.
  • Associates
  • Source and destination IP addresses
  • A pair of UDP ports one for RTP, one for RTCP

97
Real-time Transport Protocol
Time of last payload sample
Detect packet loss
Supports mixinge.g. for audio conferencing
Media Content
98
Real-time Transport Protocol
Packet loss
Source
Out-of-order recovery
Network
Destination
Late packet loss
Buffer
Playback
99
Network Issues and Design
100
Module Overview
  • Network Issues and Design
  • Quality of Service
  • Packet loss, Latency, Jitter
  • Managing Quality of Service
  • Network quality
  • MPLS
  • Silence suppression
  • Local vs. long distance
  • Network design
  • Firewalls NAT STUN

101
Quality of Service
  • QoS Definition
  • Probability of the network meeting a given
    traffic contract
  • Informally refers to the probability of a packet
    succeeding in passing between two points in the
    network within its desired latency period

102
Quality of Service
  • What can go wrong as packets go from A to B?
  • Dropped packets routers might fail to deliver
    (drop) some packets if they arrive when their
    buffers are full. Some, none, or all of the
    packets might be dropped, depending on the state
    of the network, and it is impossible to determine
    what happened in advance.
  • Delay it may take a long time for a packet to
    get from A to B because it gets held up in
    long queues or takes a less direct route to avoid
    congestion
  • Jitter packets from source will reach the
    destination with different delays, sometimes by
    taking different routes
  • Out-of-order delivery different packets may take
    different routes with enough difference in delay
    to change the order of arrival
  • Error sometimes packets are misdirected, or
    combined together, or corrupted, while en route

103
Quality of Service
104
Quality of Service
  • QoS issues can have a major impact on real-time
    media streaming
  • Packet loss
  • Missing packet replaced by silence
  • Jitter out-of-order
  • Abrupt and unwanted variation in packet arrival
    timing
  • Arrival of packets out-of-order

Packet loss
Jitter
Original
105
Quality of Service
  • Measure QoS on routers and end-points
  • QoS tends to degrade with network size and
    congestion
  • Managing Quality of Service
  • Generously over-provision a network expensive
    and does not scale
  • Reserve network resources e.g. RSVP Resource
    Reservation Protocol
  • DiffServ Differentiated services for bulk flows
    (e.g. packets from a university)
  • Multi-Protocol Label Switching (MPLS) emulates
    some properties of a circuit-switched network
    over a packet-switched network.
  • Traffic shaping control computer network traffic
    to optimize or guarantee performance, low
    latency, and/or bandwidth. Traffic shaping deals
    with concepts of classification, queue
    disciplines, enforcing policies, congestion
    management, quality of service (QoS), and
    fairness.
  • Silence suppression

106
Quality of Service
  • QoS on Local Network
  • QoS generally not a major issue in single-site
    deployments
  • Dedicate separate switches for voice and data
    traffic
  • Provide redundant networks for voice and data
    traffic
  • Gigabit Ethernet
  • Monitor QoS
  • Alarm network outages

107
Firewalls
  • Network Address Translation (NAT) re-writing the
    source and/or destination addresses of IP packets
    as they pass through a router or firewall
  • Aka network masquerading or IP-masquerading
  • Firewalls use NAT to enable multiple hosts on a
    private network to access the Internet using a
    single public IP address
  • Common in home and SOHO routers
  • SDP addresses are not translated by NAT
  • STUN (Simple Traversal of UDP over NATs, RFC
    3489)
  • Network protocol allowing clients behind NAT to
    find its (a) public address, (b) the type of NAT
    and (c) the internet side port associated by the
    NAT with a particular local port.
  • Info is used to set up UDP communication between
    two machines both behind NAT routers

108
VoIP and Speech Recognition
109
Module Overview
  • VoIP and Speech Recognition
  • MRCP v1 v2
  • Impact of CODECs
  • Network issues packet loss, latency

110
MRCP Media Resource Control Protocol
  • MRCP v1
  • IETF protocol
  • Client control of speech resources
  • Speech recognition
  • Text-to-speech
  • MRCP structure is similar to HTTP and SIP
  • Request by client in headerbody format
  • Response by server
  • Media delivery typically via RTP
  • Widely supported by VoiceXML Platforms
  • Leverages existing W3C standards for speech
    recognition and TTS markeup

111
MRCP Media Resource Control Protocol
  • MRCP v2
  • IETF protocol to supersede MRCP v1
  • Broader client control of speech resources
  • Adds speaker verification and speaker
    identification
  • Adds recording
  • Utilizes SIPSDP to establish the media pipe

112
VoIP and Speech Recognition
  • Speech Recognition and CODECs
  • Lossless CODECs do not affect speech recognition
    accuracy
  • No loss of information
  • Lossy CODECs can affect speech recognition
    accuracy
  • Greater compression tends to cause great
    degration
  • Speech recognizers are generally very reliable
    with widely deployed CODECs
  • Mobile telephony has extensive compression
  • Speech recognition trained explicitly for mobile
    performance
  • DSR Aurora Distributed Speech Recognition
  • CODEC specialized for the requirements of speech
    recognition
  • Promoted for mobile carrier usage

113
VoIP and Speech Recognition
  • Speech Recognition and Latency
  • Speech recognition not as sensitive to latency as
    humans
  • Late packet is better than no packet
  • Speech recognizers have extensive buffers for
    non-real-time processing
  • Note excessive latency (gt1sec) can cause caller
    perceived service issues

Speech Recognizer
RTP
Result
Buffer
114
VoIP and Speech Recognition
  • Speech Recognition and Packet Loss
  • Speech recognition are sensitive to packet loss
  • ASR can use packet loss information to minimize
    error reduction

Speech Recognizer
RTP
Result
Buffer
115
VoIP and Mobile Telephony
116
Module Overview
  • VoIP and Mobile Telephony
  • Analog mobile
  • Digital mobile
  • 3G and SIP
  • IMS IP Multi-Media Subsystem Architecture

117
Mobile Telephony
  • Analog Mobile
  • Experimental systems from 1920s
  • 1G 1st Generation
  • AMPS Advanced Mobile Phone Services
  • Analog transmission
  • 1978 Trial in Chicago
  • 1979 Commercial launch in Japan
  • 1981 Commercial launch in Sweden, Norway,
    Denmark, Finland
  • 1983 Commercial launch in Chicago
  • Issues limited capacity, fraud, subscriber
    volume

118
Mobile Telephony
  • 2G 2nd Generation Mobile
  • Objectives achieved
  • Digital technology
  • Increased capacity
  • Greater security against fraud
  • Global roaming
  • Advanced services
  • Lower power smaller handsets longer battery
    life
  • Many standards evolved (examples)
  • GSM Pan-European standard that spread globally
  • CDMA Americas and parts of Asia (aka PCS)
  • PDC Japan
  • Limitations
  • Optimized for voice not suited to data

119
Mobile Telephony
  • 2.5G Stepping Stone from 2G to 3G
  • 2G system with both packet switching (for data)
    and circuit switching (for voice)
  • 2.5G is a marketing term not a standard
  • Objectives achieved
  • Re-use of much 2G infrastructure (GSM CDMA)
  • Data rate of 144 kbit/sec or better
  • Used for sending photos and much more

120
Mobile Telephony
  • 3G 3rd Generation Mobile
  • Combines high-speed mobile access with Internet
    Protocol (IP) based services
  • Covers range of network technologies
  • WCDMA, CDMA2000, UMTS, EDGE
  • Data rate 384kbps for mobile systems and 2Mbps
    for stationary systems
  • Enables video, TV, images, music, games, location
    services

121
Mobile Telephony
  • 3GPP 3rd Generation Mobile
  • Collaboration agreement (Dec-98) between ETSI
    (Europe), ARIB/TTC (Japan), CCSA (China), ATIS
    (North America) and TTA (South Korea).
  • Goal global 3G specification within the scope of
    the ITU's IMT-2000 project
  • 3GPP specifications are based on evolved GSM
    specifications
  • Now generally known as the UMTS system
  • Introduced IMS

122
Mobile Telephony
  • IMS IP Multi-Media Sub-System
  • Emerged in 3GPP Release 5 with following
    enhancements
  • Principles
  • Access independence work with fixed, mobile or
    wireless networks
  • Different network architectures implement on
    operator-selected architectures
  • Terminal and user mobility provides terminal
    mobility (roaming)
  • Extensive IP-based services offer just about any
    IP-based service. VoIP, push-to-talk over
    cellular (POC), multiparty gaming, video
    conferencing, messaging, community services,
    presence information, content sharing

123
Mobile Telephony
  • IMS is Built on SIP
  • 3GPP Variant of SIP
  • Application servers for SIP session management
  • Caller ID, call waiting, call forwarding,
    transfer, call blocking, interception,
    announcements, conferencing, voice-mail, SMS
  • CSCF Call Session Control Function and other
    functions by SIP
  • Media Resource Function (MRF) SIP end-point with
    IVR-like functionality
  • TDM-VoIP gateways to bridge to fixed and mobile
    telephony
  • Whos in control?
  • TDM dumb terminals, smart network
  • Internet VoIP smart terminals, dumb network
  • IMS dumb/smart terminals, smart network

124
Closing
125
Module Objectives
  • Go home!!

126
Further Information
  • Web sites
  • IETF http//www.ietf.org/html.charters/sip-charte
    r.html
  • SIP Tutorial http//www.iptel.org/sip/siptutorial
    .pdf
  • SIP Home Page http//www.cs.columbia.edu/sip/
  • SIP Forum http//www.sipforum.org/
  • Asterisk PBX http//www.asterisk.org/
  • VoIP Wiki Reference http//www.voip-info.org/wiki
    /
  • SIP Knowledge http//www.sipknowledge.com/SIP_RFC
    .htm
  • SIP FAQ http//www.sipknowledge.com/faq_main.htm
  • SIP Tech Portal http//www.tech-invite.com/

127
Thank you!!!
About PowerShow.com