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!!!