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2G1325/2G5564 Practical Voice Over IP (VoIP):SIP and related
protocols
Spring 2006, Period 4
Maguire Cover.fm Total pages: [email protected] 2006.03.12
© 2004, 2005, 2006 G.Q.Maguire Jr. .All rights reserved. No part
of this course may be reproduced, stored in a retrieval system, or
transmitted, in any form or by any means, electronic, mechanical,
photocopying, recording, or otherwise, without written permission
of the author.
Last modified: 2006.03.12:13:16
KTH Information andCommunication Technology
Lecture notes of G. Q. Maguire Jr.
For use in conjunction with the two books:
• Luan Dang, Cullen Jennings, and David Kelly,Practical VoIP:
Using VOCAL, O’Reilly, 2002, ISBN0-596-00078-2
• Henry Sinnreich and Alan B. Johnston, InternetCommunications
Using SIP: Delivering VoIP andMultimedia Services with Session
Initiation Protocol,Wiley, 2001, ISBN: 0-471-41399-2.
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Module 1:
Introduction.........................................Welcome to the
course! ......................................Staff Associated with
the Course.........................Instructor (Kursansvarig) - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - Administrative Assistant: recording of grades,
registration, etc. - - - - - - - - - - - - - -
Goals, Scope and Method...................................Goals
of the Course - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - -Scope and Method - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - -
Prerequisites........................................................Contents
..............................................................Topics
.................................................................Examination
requirements ..................................Project
.................................................................Assignment
Registration and Report
..................Literature.............................................................Lecture
Plan........................................................Voice
over IP (VoIP)
............................................Potential Networks
..............................................Internetworking....................................................
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VoIP a major
market............................................Handsets.............................................................VoIP
Chipsets
.....................................................Deregulation⇒
New operators ...........................Deregulation⇒ New
Suppliers...........................Let them fail
fast!.................................................Latency
...............................................................VOIP
Modes of Operation...................................IP based
data+voice infrastructure .....................Voice
Gateway....................................................Voice
over IP (VOIP) Gateways ..........................Voice
representation - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - -Signaling - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - -Fax Support - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
Management - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - Compatibility - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - -
Cisco’s Voice Over IP
.........................................Intranet Telephone System
.................................Wireless
LANs.....................................................
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Telia’s
HomeRun.................................................Ericsson’s
"GSM on the Net" ..............................VOIP vs. traditional
telephony .............................Economics
..........................................................VoIP vs.
traditional telephony
..............................Patents................................................................Deregulation⇒
Trends .......................................Carriers offering
VOIP .........................................MCI (formerly
WorldCom) Connection................Previously - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - Today - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
Level 3 Communications
Inc...............................TeliaSonera
Bredbandstelefoni...........................Emulating the
PSTN............................................Calling and Called
Features................................Beyond the PSTN: Presence
& Instant MessaginPresence-Enabled Services
................................Three major alternatives for VoIP
.......................
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Negatives
............................................................Deregulation⇒
New Regulations.......................Regulations in Sweden
.......................................Programmable “phone”
.......................................Conferences
........................................................Not with
out problems .........................................References
and Further
Reading........................Acknowledgements.............................................Module
2: VoIP details........................................Traditional
Telecom vs. Datacom........................VoIP details: Protocols
and Packets ...................RTP and H.323 for IP Telephony
.......................RTP, RTCP, and
RTSP.......................................Real-Time Delivery
.............................................Packet
delay........................................................Dealing
with Delay jitter ......................................Delay and
delay variance (jitter)..........................
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Playout delay
......................................................When to
play.......................................................Retransmission,
Loss, and Recovery ..................Patterns of Loss
..................................................Loss
concealment................................................VoIP
need not be “toll quality” .............................RTP:
Real-Time Transport Protocol....................Payload
types......................................................Audio
Encodings
.................................................Timestamps.........................................................Stream
translation and mixing.............................RTP Control
Protocol (RTCP) ...........................Compound Reports
.............................................Proposed RTCP
Reporting Extensions...............RTP translators/mixers
.......................................Synchronizing Multiple
Streams ..........................RTP Transport and Many-to-many
Transmission
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Sessions, Streams, Protocol Port, and DemultiplFurther details
of RTP and RTCP........................Real Time Streaming Protocol
(RTSP) ...............RTSP session description
...................................References and Further
Reading........................RTP and RTCP - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - RTSP- - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - -
Module 3:
SIP......................................................Session
Initiation Protocol (SIP) .........................SIP WG’s
deliverables.........................................Related
working groups.......................................Historic - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - -
Session Initiation Protocol (SIP) .........................Is
SIP simple?
.....................................................SIP, RTP, and
RTSP...........................................SIP actors
............................................................SIP
Methods and Status Codes..........................SIP Status codes
- patterned on and simular to HTTP’s status codes: - - - - - - - -
- -
SIP Uniform Resource Indicators (URIs).............
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Issues to be considered
......................................Address
Resolution.............................................SIP timeline
.........................................................SIP Invite
.............................................................Bob’s
response to Alice’s
INVITE........................ACK.....................................................................SIP
Invite (method/URI/version)..........................SIP
Via................................................................Dialog
(Call leg) Information ................................SIP
CSeq.............................................................SIP
Contact
.........................................................SIP
Content Type and Length ............................SIP
Max-Forwards...............................................Other
header fields..............................................Several
types of SIP Servers..............................SIP Trapezoid
....................................................SIP Call
Setup.....................................................
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SIP Call Setup
Attempt........................................SIP Call Setup
Attempt........................................SIP Presence
......................................................SIP B not
Present ................................................SIP
Registration Example...................................Purpose of
registration........................................REGISTERing
......................................................SIP Call
Setup Attempt........................................SIP Session
Termination using BYE...................SIP Session Termination
using CANCEL............CANCEL andOPTIONS
............................ 163CANCEL - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - 163OPTIONS - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
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Unsuccessful final responses are hop-by-hop.....Authentication
.....................................................SIP Method
Extensions in other RFCs ................SIP Extensions and
Features..............................SIP Presence - Signed
In....................................
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SUBSCRIBE andNOTIFY .......................... 169SIP Instant
Messaging Example .........................SIP Instant Messaging
Example (continued).......Message
example...............................................Midcall
signalling .................................................Call
Control
.........................................................Example of
usingREFER ........................... 175QoS and Call
Setup.............................................SIP Message
retransmission ..............................RFC 3261 - Routing
Changes.............................RFC 3261 - New Services
..................................Compression of SIP
............................................Intelligent Network
service using SIP ..................Capability Set 1:
Services...................................Capability Set 2
...................................................Features..............................................................SIP
development, evolution, …...........................
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- - - - - 198
) ...............................
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Gateways.............................................................Significance
.........................................................References
and Further Reading........................SIP - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - -ITU Services CS-1 and CS-2 - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
Module 4: Session Annoucement Protocol (SAPSession Announcement
Protocol (SAP) .............References and Further
Reading........................SAP- - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - -
Module 5: Session Description Protocol (SDP) ..Session
Description Protocol (SDP)....................Session Description
Protocol (SDP)....................Internet drafts related to SDP: -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - -
SDP Message
Details..........................................Session
description .............................................SDP
Offer/Response Example............................SDP Response
Example ....................................Session Modification
...........................................Session modification
(continued) ........................
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- - - - - - - - - - - - 219
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Start and Stop
Times...........................................Grouping of Media
Lines in the Session DescriptLip Synchronization
............................................Next generation of SDP
(SDPng) ........................SDPng
structure..................................................Why
XML?...........................................................References
and Further Reading........................SDP- - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - -
Module 6: DNS and ENUM
.................................Telephony URL and Phone-Context
...................SIP URL
..............................................................ENUM
.................................................................DNS
....................................................................NAPTR
- Naming Authority Pointer [77] ..............To find the DNS names
for a specific E.164 numENUM
Services...................................................EUNM
Timeline ...................................................
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244................................... 245 - - - - - - - - - - - -
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- - - - 245
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Sweden’s ENUM Mapping...................................VISIONng
Association.........................................SIP goes beyond
ENUM......................................References and Further
Reading........................DNS - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - -ENUM- - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - -
Module 7: SIP Mobility
........................................SIP Mobility
.........................................................Local
Number Portability .....................................References
and Further Reading........................SIP Mobility - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - -Service Mobility - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
-
Module 8: SIP Service Creation ..........................SIP
Service Creation...........................................Services
implemented by x.................................Services
implemented by Extensions .................SIP Service Logic
...............................................Call Processing
Language (CPL)........................SIP Common Gateway Interface
(CGI)...............
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................................. 254gy - SIP and Jain .....
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260™)..........................
261................................... 262 - - - - - - - - - - - -
- - - - - - - - - - - - - 262 - - - - - - - - - - - - - - - - - - -
- - - - - 263- - - - - - - - - - - - - - - - - - - - - - - - - 263-
- - - - - - - - - - - - - - - - - - - - - - - - 263 - - - - - - - -
- - - - - - - - - - - - - - - - 263- - - - - - - - - - - - - - - -
- - - - - - - - 264
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SIP Java Servlets
...............................................JAIN
APIs............................................................US
National Institute of Standards and TechnoloParlay
..................................................................SIP
Request-URIs for Service Control ................Reason Header
..................................................Voice eXtensible
Markup Language (VoiceXMLReferences and Further
Reading........................SIP Service Creation - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- -JAIN - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - -Parley - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - SIP Request URI - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - Reason Header - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - -VoiceXML - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - -
Module 9: User Preferences................................User
Preferences ................................................Contact
parameters ............................................Contact
header
example.....................................Accept/Reject-Contact
header(s) ........................
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280................................. 281 - - - - - - - - - - - - -
- - - - - - - - - - - - 287
TESLA) .................. 289.................................
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Callee (i.e., called party) Parameter processing .Accept-Contact
Example.....................................Request-Disposition............................................SIP
Service
Examples.........................................Privacy-Conscious
Personalization ....................References and Further
Reading........................User Preferences - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - -
Module 10: SIP Security, NATs, and Firewalls ...SIP Security
........................................................SIP Digest
Authentication ...................................SIPand S/MIME
..................................................SDP & RTP
security............................................Secure Call
Setup [126] - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - -
Efficient Stream Loss-tolerant Authentication (NATs and
Firewalls.............................................Types of
NAT......................................................Cone vs.
Symmetric NAT ...................................NAT traversal
methods........................................
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Network Address Translation))
..................................
297.................................
298................................. 299ll Traversal ...........
300..............................
301................................ 302ASN.1) ...................
303................................
305.................................
306...............................
307................................
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STUN (Simple Traversal of UDP through NATs (296STUN
steps.........................................................UDP
and TCP Firewall Traversal problems........UDP and TCP NAT
Traversal problems.............SIP Application Level Gateway (ALG)
for FirewaMiddlebox communications (MIDCOM)
..............Application aware Middlebox
..............................Security flaws in Abstract Syntax
Notation One (Swedish Electronic Communications Act
............Recording of Call Contents
.................................Privacy & Lawful Intercept
(LI)...........................Reasonably Available
Information.......................EU privacy and Lawful Intercept
(LI) ...................Intercept architecture
..........................................Voice over IP Security
Alliance ...........................Spam over Internet Telephony
(SPIT).................
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- - - - - - - - - - - - 313- - - - - - - - - - - - - - - - - - - -
- - - - - 314- - - - - - - - - - - - - - - - - - - - - - - - - 316
- - - - - - - - - - - - - - - - - - - - - - - - 319
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32875]...........................
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333................................. 334EOPRIV)................
335................................... 336 - - - - - - - - - - - -
- - - - - - - - - - - - - 336
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References and Further Reading........................SIP
Security - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - RTP encryption - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - NATs and Firewalls - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Privacy
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - -
Module 11: SIP Telephony..................................SIP
Telephony
....................................................Telephony
Routing over IP (TRIP) ......................Call Control
Services...........................................Call Center
Redesign using SIP..........................Additional SIP
Telephony services .....................Emergency Telecommunication
Service (ETS)[1Emergency Services (E911)
...............................Public Safety Answering Point
(PSAP)................Vonage 911
service.............................................Vonage equips
PSAps with VoIP........................Geographic Location/Privacy
Working Group (GReferences and Further
Reading........................Emergency services - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- -
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- - - - - - - - - - - - - - - - - - - 337
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342..................................
343................................... 344- - - - - - - - - - - - -
- - - - - - - - - - - - 344- - - - - - - - - - - - - - - - - - - -
- - - - - - 345 - - - - - - - - - - - - - - - - - - - - - - - -
345- - - - - - - - - - - - - - - - - - - - - - - - - - 345
................................ 347
................................. 348
................................ 349et Servers (SPIRITS)
350................................
351................................... 352 - - - - - - - - - - - -
- - - - - - - - - - - - 352 - - - - - - - - - - - - - - - - - - - -
- - - - 352 - - - - - - - - - - - - - - - - - - - - - - - - 354
[email protected] 2006.03.12
SIP Telephony - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - TRIP - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - -
Module 12: SIP Conferencing
.............................Conferencing.......................................................Conferencing
Models [176].................................SIP
Conferencing................................................Speaker
recognition in a conference...................References and
Further Reading........................SIP Conferencing - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - Session Annoucement Protocol - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - SMIL- - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - -Speaker recognition in a conference - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
Module 13: Mixed Internet-PSTN Services .........Mixed
Internet-PSTN Services............................PSTN and
Internetworking (PINT) ......................Servers in the PSTN
Initiating Requests to InternTelephony Routing over IP (TRIP)
......................References and Further
Reading........................PINT - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - -SPIRITS - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - -TRIP - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - -
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- - - - - - - - - - - - - - - - - - - - - - - - 354
................................
355.............................
356.................................
357..................................
358.................................
359...................................
360................................
361.................................
362................................
364................................... 365- - - - - - - - - - - - -
- - - - - - - - - - - - 365- - - - - - - - - - - - - - - - - - - -
- - - - - 365
..................................
366................................
367..................................
368.................................
369.................................. 370
[email protected] 2006.03.12
ISUP - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - -
Module 14: AAA and QoS for
SIP.......................Authentication, Authorization, Accounting
(AAA)SIP
Accounting....................................................Open
Settlement Protocol (OSP)........................Achieving QoS
....................................................Some measured
delays.......................................Underlying Quality
...............................................Voice
Quality.......................................................Rating
voice quality in practice ............................QoS
Proprietary vs. Standards based.................Past - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - 2002 - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - -
QoS for
SIP.........................................................VoIP
traffic and Congestion Control....................Delay and Packet
Loss effects ............................When to continue (try
again) ...............................More about congestion
.......................................
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20 of 22Practical Voice Over IP (VoIP): SIP and related
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................................ 371
................................ 372
...................................
373................................ 379 (SIPPING) .............
380.................................
382..................................
383............................... 384- - - - - - - - - - - - - - -
- - - - - - - - - - 385 - - - - - - - - - - - - - - - - - - - - - -
- - - 386
ponse (IVR) ............ 387..................................
388..................................
389................................... 390 - - - - - - - - - - - -
- - - - - - - - - - - - 390
................................ 391
..................................
392................................. 393
[email protected] 2006.03.12
VoIP quality over IEEE
802.11b..........................Application Policy Server
(APS)..........................References and Further
Reading........................Module 15: SIP Applications
...............................Session Initiation Protocol Project
INvestiGationApplication Service Components
........................Advantages
.........................................................Collecting
DTMF digits for use within a service ..Reponse “3. 200 OK” looks
like: - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - Controller issues a “re-Invite” at 11 which looks like: -
- - - - - - - - - - - - - - - - - - - - -
Voice Portal Service using Interactive Voice ResManaging
Services..............................................Lots more
services ..............................................References
and Further Reading........................SIPPING - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - -
Module 16: More than
Voice................................Non-voice Services and IP
Phones.....................XML
....................................................................
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.................................
394..................................
395.................................
396.................................
397..................................
398................................
399..................................
400................................
402.................................. 405- - - - - - - - - - - - -
- - - - - - - - - - - - 405 - - - - - - - - - - - - - - - - - - - -
- - - - 405
................................... 406 - - - - - - - - - - - -
- - - - - - - - - - - - - 406- - - - - - - - - - - - - - - - - - -
- - - - - - 406
................................ 409
................................ 410
.................................
411................................
412................................. 413
[email protected] 2006.03.12
Invoking RTP streams
.........................................More details
........................................................Services
for sale - building a market ...................Network Appliances
............................................Proposed Extension of
SIP .................................Service Location Protocol
(SLP) URL.................Example
service..................................................Example of
service portability..............................Text
.....................................................................Interleaved
text - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - Timed Text- - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - -
References and Further Reading........................Phone
Services - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - - - - - - - - - - - - - - -Network Appliances- - - - - -
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - -
Module 17:
VOCAL.............................................VOCAL System
Overview...................................VOCAL
Servers...................................................Scaling
of a VOCAL system ................................For comparison
with a PBX ................................
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22 of 22Practical Voice Over IP (VoIP): SIP and related
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.................................
414..................................
415...................................
416.................................
417...................................
418are............................
419..................................
420..................................
421..................................
422..................................
423..................................
424...................................
426.................................
427..................................
428................................... 429
[email protected] 2006.03.12
Marshal server
(MS)............................................Redirect Server
(RS)...........................................Feature Server
(FS)............................................Residential Gateway
(RG)...................................References and Further
Reading........................Module 18: SIP Express Router and
other SoftwSIP Express Router
(SER)..................................SipFoundry
.........................................................Other SIP
Proxies ...............................................SIP Tools
............................................................SIP
Clients
..........................................................References
and Further Reading........................Module 19: Non-SIP
applications .......................Skype
..................................................................References
and Further Reading........................
-
2G132 oice Over IP (VoIP):rotocolsriod 4
Maguire Total pages: [email protected]
© 2004, 2005, 2006 G.Q.Maguire Jr. .All rights reserved. No part
of this course hotocopying, recording, or otherwise, without
written permission of the author.
Last modified: 2006.03.12:16:45
KTH Information andCommunication Technology
Introductionof G. Q. Maguire Jr.
the two books:
nings, and David Kelly,OCAL, O’Reilly, 2002, ISBN
an B. Johnston, Internet SIP: Delivering VoIP andth Session
Initiation Protocol,71-41399-2.
5/2G5564 Practical VSIP and related p
Spring 2006, Pe
Introduction.fm 2006.03.12
may be reproduced, stored in a retrieval system, or transmitted,
in any form or by any means, electronic, mechanical, p
Module 1:Lecture notes
For use in conjunction with
• Luan Dang, Cullen JenPractical VoIP: Using V0-596-00078-2
• Henry Sinnreich and AlCommunications UsingMultimedia Services
wiWiley, 2001, ISBN: 0-4
-
Module 1: 24 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
se!
s on SIP and related protocols,ls which are used.
e course web page
ation of KTH to:
Maguire Welcome to the [email protected] 2006.03.12
Welcome to the courThe course should befun.
We will dig deeper into Voice over IP - with a focubut may also
examine some of the other protoco
Information about the course is available from
thhttp://www.imit.kth.se/courses/2G1325/
Note that the above URL will change - due to the reoganiz
http://www.cos.ict.kth.se/education/msc/ccs/courses/2G1325/
http://www.imit.kth.se/courses/2G1325/http://www.cos.ict.kth.se/education/msc/ccs/courses/2G1325/
-
Module 1: 25 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
Course
se>
Maguire Staff Associated with the [email protected]
2006.03.12
Staff Associated with the Instructor (Kursansvarig)
prof. Gerald Q. Maguire Jr.
Administrative Assistant: recording of grades, registration,
etc.
Irina Radulescu
-
Module 1: 26 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
thod
ms are, their basic
re.development in this area.
mmunication Application
nd examine some places where it can explore).
g a written report andject.
Maguire Goals, Scope and [email protected] 2006.03.12
Goals, Scope and MeGoals of the Course
• To understand what Voice over IP (VoIP) systearchitectures,
and the underlying protocols
• To be able to read and understand the literatu• To provide a
basis for your own research and Scope and Method
• We are going to examine the Vovida Open CoLibrary (VOCAL )
system1• to understand both the details of the system(s) and• to
abstract from these details some architectural features a
can be extended (thus using it as a platform on which you
• You will demonstrate your knowledge by writingiving an oral
presentation describing your pro
1. The source code is available fromvovida.org
vovida.org
-
Module 1: 27 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
ations (this requires
Maguire [email protected] 2006.03.12
Prerequisites• Internetwork (2G1305) or• Equivalent knowledge in
Computer Communic
permission of the instructor)
-
Module 1: 28 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
(VoIP) systems are, their basicwill primarily focus on
theocols.
a project of ~50 hours effort.
Maguire [email protected] 2006.03.12
ContentsThe focus of the course is on what Voice over
IParchitectures, and the underlying protocols. We Session
Initiation Protocol (SIP) and related prot
The course consists of 10 hours of lectures and
-
Module 1: 29 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
erver, Registrar Server,
Maguire [email protected] 2006.03.12
Topics• Session Initiation Protocol (SIP)• Real-time Transport
Protocol (RTP)• Real-time Streaming Protocol (RTSP)• Common Open
Policy Server (COPS)• SIP User Agents• Location Server, Redirect
Server, SIP Proxy S
... , Provisioning Server, Feature Server• Call Processing
Language (CPL)
-
Module 1: 30 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
ents
Maguire Examination [email protected] 2006.03.12
Examination requirem• Written and Oral project reports
Grades: U, 3, 4, 5
-
Module 1: 31 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
and to show that you haveourage you to find a topic which
y understand the material)
med by yourself). Each oral reports.ctor before starting.
Maguire [email protected] 2006.03.12
ProjectGoals: to gain analytical or practical experience
mastered some knowledge in this area and to encinterests you (since
this will motivate you to reall
• Can be done in a group of 1 to 3 students (forstudent must
contribute to the final written and
• Discuss your ideas about topics with the instru
-
Module 1: 32 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
d Report.se>, subject=2G1325 topic
00 words) for each student.ach paper suitable for submission
(in the case where the report is ap can be explained in the
overall
; 2) who did what; if you haved describe the methods and toolsur
analysis.
resentations scheduled
.se>
the deadlines!
Maguire Assignment Registration and [email protected]
2006.03.12
Assignment Registration an• Registration: 3 April 2006, to
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Module 1: 33 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
s:
Practical VoIP: Using
t Communications Using with Session Initiation
s necessary. A list of interestingand in the references and
further
rature in conjunction with youre your sourcesin your report.
Maguire [email protected] 2006.03.12
LiteratureThe course will mainly be based on the two book
• Luan Dang, Cullen Jennings, and David Kelly,VOCAL, O’Reilly,
2002, ISBN 0-596-00078-2.
• Henry Sinnreich and Alan B. Johnston, InterneSIP: Delivering
VoIP and Multimedia ServicesProtocol, Wiley, 2001, ISBN:
0-471-41399-2
We will refer to other books, articles, and RFCs aliterature
will be available on the course web pagereading section of each
lecture module.
In addition, you will be searching & reading the
liteprojects. Please make sure that youproperly referenc
-
Module 1: 34 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
Maguire Lecture [email protected] 2006.03.12
Lecture Plan• Introduction
• Course arrangement• Set the context of VoIP, both technically
and economically
• VoIP details• Session Initiation Protocol (SIP)• Session
Description Protocol (SDP)• DNS and ENUM
• Mobility• Service Creation• User preferences• Security, NATs,
and Firewalls• SIP Telephony• Conferencing• Mixed Internet - PSTN
services• AAA and QoS• More than just voice!
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Module 1: 35 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
)srocessingin theend points.
etwork - where processing is
(Now) we think about aconverged network whichis aglobal
network
k
t
terminal
Maguire Voice over IP (VoIP)[email protected] 2006.03.12
Voice over IP (VoIPVoIP is an End-to-End Architecture which
exploitp
Unlike the traditional Public Switch Telephony Ndoneinside the
network.
Network Convergence:In the past, many different networks -each
optimized for aspecific use: POTS, data networks (such as X.25),
broadcastradio and television, … and each of these in turn often
hadspecific national, regional, or proprietary implementations)
⇒
CODEC
IP stack
radio
CODEC
IP stac
etherne
Cellular IP terminal Fixed IP
VoIP server
call/sessionroutingtranscoding
IP cloud
IP end-to-end
-
Module 1: 36 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
s
f the underlying network,
Ring
MH
Ad hoc
AN
UPSTN
Maguire Potential [email protected] 2006.03.12
Potential Network
• We will focus on VoIP, largely independently oi.e., LAN,
Cellular, WLAN, PAN, Ad hoc, … .
Figure 1: Internet and PSTN
WANswitch
switch switch
switchR
R
R
R
R
R
FDDI
Token H
H
MSCHLR/VLR
BSCBTS
IWU
Cellular networks
WLAN
P
Ethernet LANs
AP
… …
…
…
IW
MH
MH
MH
MH
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Module 1: 37 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
of multiple networksechnologies by providing and makes them
aed sampling rate, typicallyvoice coding; however, VoIP isr or
lower data rates bandwidth between the end
fers is quality which is:
still acceptable)
systems.
Maguire [email protected] 2006.03.12
InternetworkingInternetworking is
• based on the interconnection (concatenation)• accommodates
multiple underlying hardware t
a way to interconnect heterogeneous networksinter-operate.
Public Switched Telephony System (PSTN) usesfix8kHz and coding
to 8 bits, this results in 64 kbpsnot limited to using this coding
and could havehighedepending on the CODEC(s) used, the
availablepoints, and the user’s preference(s).
One of the interesting possibilities which VoIP of
• better that “toll grade” telephony or• worse than “toll grade”
telephony (but perhaps
This is unlike thefixed quality of traditional phone
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Module 1: 38 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
etwhich began with H.323 and hasrs of users and a large variety
ofincreasing numbers of vendors,arket?
tions around 1997, buts that it took more thans, but the next 1
million
er Cisco losing momentum?”,tember 17, 2003, 4:00 AM PT
5), they had shipped theiran one year earlier.)
Maguire VoIP a major [email protected] 2006.03.12
VoIP a major markVoice over IP has developed as a major market
-now moved to SIP. There are increasing numbeVoIP hardware and
software on the market. Withthe competition is heating up - is it a
maturing m
“Cisco began selling its VoIP gear to corporauntil the past
year, sales were slow. Cisco notethree years to sell its first 1
million VoIP phonetook only 12 months.”
Ben Charny , “Is VoIP pioneCNET News.com, Sep
As of their fiscal year 2005 (ending July 30, 2006 millionth IP
phone[9]. (This is 3 million more th
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Module 1: 39 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
ets:
.clarisys.com/ ,k, IPEVO, …
® Data Phone.com/rverndset and their server, but I this
mis-feature.
nd e805) with VoIP and.
ml
d GSM & 802.11 ("GSM.11"),
Maguire [email protected] 2006.03.12
HandsetsThere are now lots of USB attached VoIP hands
• http://www.clarisys.com/ Claritel-i750 http://wwwPlantronics
DA60, Net2Phone, Linksys, D-Lin
WLAN Handsets:
• starting with Symbol Technologies’s NetVision• Vocera
Communications Badge http://www.vocera
• runs speech recognition software in a network attached se•
unfortunately it uses a proprietary protocol between the ha
expect others will make similar devices which will not have
• Toshiba announced two PDAs (models e800 aWi-Fi; as have NTT’s
DoCoMo and Sony Corp
• for more http://www.sipcenter.com/vsts/vsts_sipphones.ht
VoIP cellular handsets, e.g. TTPCom’s combineNokia, Qtek, …
http://www.vocera.com/http://www.sipcenter.com/vsts/vsts_sipphones.htmlhttp://www.clarisys.com/
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Module 1: 40 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
issinesstelephonehandsetsutions.html
, IrDA, SPI, 16 programmable I/O, …lumn inputs/outputs to
control up …
or core running at 80 MIPS compression/decompression
og-to-digital converters (A/D),t and output)
nable high quality
Maguire VoIP [email protected] 2006.03.12
VoIP ChipsetsAgere Systems’ VoIP Phone-On-A-Chip -
targetbuandspeakerphoneshttp://www.agere.com/mobility/voip_sol
Two ICs:
• T8302 IPT_ARM (Advanced RISC Machine)• Up to 57.6 MHz
general-purpose processor• controls the system I/O: two
10/100Base-T Ethernets, USB
pins (some could be used to interface to an LCD module)• general
telephone control features: 7 row outputs and 8 co
to 56 LEDs and scan up to 56 keys, 6 different flash rates,
• T8301 IPT_DSP (digital signal processor)• Based on Agere
Systems DSP1627 digital signal process• single-cycle multiply
accumulate instruction supports voice
and echo cancellation algorithms• Includes two 16-bit
digital-to-analog (D/A), one 16-bit anal
low-pass filters, audio amplifier, lots of buffers (for for
inpu
A special feature isacoustic echo cancellation to espeakerphone.
See also [3].
http://www.agere.com/mobility/voip_solutions.html
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Module 1: 41 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
erators
ch technology
loads as of 2005.02.10)
7,596,653 as of 2005.02.10
rss.xml updated
Maguire Deregulation ⇒ New [email protected]
2006.03.12
Deregulation ⇒ New opLots of new actors appeared as
operators:
• MCI (formerly Worldcom) - http://www.mci.com/• Qwest -
http://www.qwest.com/• Level3 http://www.level3.net/
• (3)Voice, an IP based long distance service using Softswit
• Vonage - http://www.vonage.com /• 1.5 million lines in
service• > 42 million as of March 1, 2006 [12]
• Skype™ Technologies http://www.skype.com/• “Skype is free
Internet telephony that just works.”• 200 million downloads as of
2005.11.08 (67,430,762 down
– ~1 Million downloads/day– downloads at peak are ~0.5
Gbit/sec
• 19,627,534,145 minutes served as of 2006.03.12 vs. 4,70• >
5 million simultaneous users on Jan. 23 2006 [13]• statistics as an
RSS feed at: http://share.skype.com/stats_
every few minutes
• …
http://www.skype.com/http://www.vonage.comhttp://www.level3.net/http://www.qwest.com/http://www.mci.com/http://share.skype.com/stats_rss.xml
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Module 1: 42 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
uppliers
atacom vendors.
vendors.
o opertors) were reorganizing,to the Telecom meltdown!
ucts/sip/
Maguire Deregulation ⇒ New [email protected]
2006.03.12
Deregulation ⇒ New SLots of new actors as equipment
suppliers:
• Cisco , 3Com, Nortel Networks, …
Traditional telecom equipment vendors buying d
Lots of mergers and acquisitions among datacom
As of Fall 2002, many of these vendors (similar tselling off
divisions, reducing staffing, … -- due However, some have survived
(or been reborn).
For a list of SIP products see:http://www.pulver.com/prod
http://www.pulver.com/products/sip/
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Module 1: 43 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
lecom troubles is thatas subsumed (and will
embodied in otherausing the immediateircuit-based telephony
atson, Robert J. Berger, et al.,hael Powell, October 21,
2002press/powell.html
which recommends that the FCC:ts. The policy should be one
of
t, while not yet extinct, is artificially prolong its use.
to thwart municipal,on’t fit the telephone company
d aggressively expand unlicensed
Maguire Let them fail [email protected] 2006.03.12
Let them fail fast!We hold that the primary cause of current
teInternet-based end-to-end data networking hsubsume) the value
that was formerlycommunications networks. This, in turn, is
cobsolescence of the vertically integrated, cindustry of 127 years
vintage.
Izumi Aizu, Jay BLetter to FCC Chairman Mic
http://pulver.com/
The extent of this transformation is well described in their
complete letter• ‘‘Resist at all costs the telephone industry’s
calls for bailou
"fast failure."• Acknowledge that non-Internet communications
equipmen
economically obsolete and forbear from actions that would•
Discourage attempts by incumbent telephone companies
publicly-owned and other communications initiatives that
dbusiness model.
• Accelerate FCC exploration of innovative spectrum use
anspectrum allocation.’’
http://pulver.com/press/powell.htmlhttp://www.pulver.com/products/sip/
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Module 1: 44 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
elay (adapted from a drawing by Ciscoa)
avg (ms) max (ms) hops
0 4 221 21 21 128 39 44 1422 124 135 1599 199 200 2350 350 351
20
800 900 ms
elephony
Maguire [email protected] 2006.03.12
Latency
For example:
Figure 2: Usability of a voice circuit as a function of
end-to-end da. (this was at
http://www.packeteer.com/solutions/voip/sld006.htm)
Round-trip times from dumburken.it.kth.se(as of 2004.03.26)
min (ms)
Local LANs (www.imit.kth.se) 0to northern Sweden
(cdt-lisa.cdt.luth.se)to Austria (www.tu-graz.ac.at) 3To my machine
in eastern US (via an SDSL link) 1To US west coast
(www.stanford.edu) 1To Australia (www.uow.edu.au) {via the US west
coast} 3
Usability
1
0100 200 300 400 500 600 700
Toll quality Satellite CB Radio
FAX relay/broadcast
Internet t(past)(now!)
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Module 1: 45 of 84Practical Voice Over IP (VoIP): SIP and
related protocols
tion
an operator
al/national network to the same
Maguire VOIP Modes of [email protected] 2006.03.12
VOIP Modes of Opera• PC to PC• PC-to-Telephone calls•
Telephone-to-PC calls• Telephone-to-Telephone calls via the
Internet• Premises to Premises
• use IP to tunnel from one PBX/Exchange to another• see Time
Warner’s “Telecom One Solution”
• Premises to Network• use IP to tunnel from one PBX/Exchange to
a gateway of
• Network to Network• from one operator to another or from one
operator’s region
operator in another region or nation
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tructure
AP Mobile
In-building WLAN system
Handset
Maguire IP based data+voice [email protected]
2006.03.12
IP based data+voice infras
exchange
Workstation
E-1 to PSTN AP
GatewayWorkstation
Router
Internet
Gateway
Voice
Handset
Handset
Public cells
Home
Office
FW/Switch
IP Phone
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Module 1: 47 of 84Practical Voice Over IP (VoIP): SIP and
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urrently (formerly?) used foror special purpose gateways
such
LAN
LANadaptor
Maguire Voice [email protected] 2006.03.12
Voice Gateway
Use access servers filled with digital modems (ccurrent analog
modem pools) as voice gatewaysas that of Li Wei [4].
CPU
Modem chip
2B+D or 30B+D or …
digital pathISDN interface
A/D converterD/A converter
Digitized voice or data
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Module 1: 48 of 84Practical Voice Over IP (VoIP): SIP and
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tewaysfax services, but can also enabletegrated messaging,
least-cost
ce, encapsulate it into packets, and send itpressed, buffered,
and then sent out as
world and the E.164 telephone numbering
ding/decoding or G.729
Maguire Voice over IP (VOIP) [email protected]
2006.03.12
Voice over IP (VOIP) GaGateways not only provide basic telephony
and lots of value-added services, e.g., call-centers, inrouting, …
.
Such gateways provide three basic functions:• Interface between
the PSTN network and the Internet
Terminate incoming synchronous voice calls, compress the voias
IP packets. Incoming IP voice packets are unpacked,
decomsynchronous voice to the PSTN connection.
• Global directory mappingTranslate between the names and IP
addresses of the Internetscheme of the PSTN network.
• Authentication and billing
Voice representation
Commonly: ITU G.723.1 algorithm for voice enco(CS-ACELP voice
compression).
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Module 1: 49 of 84Practical Voice Over IP (VoIP): SIP and
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ventional signaling will be used
ly happens at thebeginningor what can be enabled via SIP
.
ntire FAX before
anagement Information Base)
yropped calls, lost/resent
Maguire Voice over IP (VOIP) [email protected]
2006.03.12
Signaling
Based on the H.323 standard on the LAN and conon telephone
networks.
NB: In conventional telephony networks signallingonandendof
acall. See Theo Kanter’s dissertation fso that you can react
toother events.Fax Support
Both store-and-forward and real-time fax modes
• In store-and-forward the system records the etransmission.
Management
Full SNMP management capabilities via MIBs (M
• provided to control all functions of the Gatewa• Extensive
statistical data will be collected on d
packets, and network delays.
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ch simpler than H.323
Maguire Voice over IP (VOIP) [email protected]
2006.03.12
Compatibility
De jure standards:• ITU G 723.1/G.729 and H.323• VoIP Forum IA
1.0
De facto standards:• Netscape’s Cooltalk• Microsoft’s NetMeeting
(formerly H.323, now SIP)
Session Initiation Protocol (SIP) [RFC 2543] is mu
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IPoice traffic (e.g., telephone calls
rrying packetized audio and
Maguire Cisco’s Voice Over [email protected] 2006.03.12
Cisco’s Voice Over Enables Cisco 3600 series routers to carry
live vand faxes) over an IP network.
They state that this could be used for:
• Toll bypass• Remote PBX presence over WANs• Unified voice/data
trunking• POTS-Internet telephony gateways
Uses Real-Time Transport Protocol (RTP) for cavideo traffic over
an IP network.
Cisco 3600 supports a selection of CODECs:
• G.711 A-Law 64,000 bits per second (bps)• G.711 u-Law 64,000
bps• G.729 8000 bps
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ly need to send traffic if thereke this because it enables
even
ey hearabsolute silence, so toise when there is nothing useful
to generate backgroundated”.
e gateway with software such as
Maguire Cisco’s Voice Over [email protected] 2006.03.12
Cisco 3800 supports even more CODECs:
• ITU G.726 standard, 32k rate• ITU G.726 standard, 24k rate•
ITU G.726 standard, 16k rate• ITU G.728 standard, 16k rate
(default)• ITU G.729 standard, 8k rate
By using Voice Activity Detection (VAD) - you onis something to
send {Note: telecom operators lihigher levels of statistical
multiplexing}.
An interesting aspect is that users worry when thhelp make them
comfortable it is useful to play noto output. Cisco provide a
“comfort-noise commandnoise to fill silent gaps during calls if VAD
is activ
Cisco 3600 series router can be used as the voicMicrosoft
NetMeeting.
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ates either current voice rate or
ed/113t/1
Maguire Cisco’s Voice Over [email protected] 2006.03.12
Cisco 3800 also supports “fax-relay” - at various
r2,400/4,800/7,200/9,600/14,400 bps fax rates.
For futher information
seehttp://www.cisco.com/univercd/cc/td/doc/product/software/ios11313t_1/voip/config.htm
http://www.cisco.com/univercd/cc/td/doc/product/software/ios113ed/113t/113t_1/voip/config.htm
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stemco Systems announced thattVision™ wireless LAN handsetocal
area network telephone
infrastructure and a voice system conforms to H.323.
rea network telephone
ming algorithm with load
eous, full-duplex phone calls.
s WebSwitch2000.
Maguire Intranet Telephone [email protected]
2006.03.12
Intranet Telephone SyOn January 19, 1998,Symbol Technologies and
Cisthey had combined the Symbol Technologies’ Neand Cisco 3600 to
provide a complete wireless lsystem based on Voice-Over-IP
technology.
The handset uses a wireless LAN (IEEE 802.11)gateway via Cisco
3600 voice/ fax modules. The
"I believe that this is the first wireless local abased on this
technology" -- Jeff Pulver
Seamless roaming via Symbol’s pre-emptive roabalancing.
Claims each cell can accommodate ~25 simultan
Ericsson partnered with Symbol, using Ericsson’
http://www.cisco.com/univercd/cc/td/doc/product/software/ios113ed/113t/113t_1/voip/config.htmhttp://www.symbol.com
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sed data solutions through thewill primarily enhance Telia
t will lead to the introduction ofdata connections are
currentlybe introduced.
ources it needs to maintain itsof radio-based LAN solutions.
edium-sized companies as well as bytions.
ducts that do not require frequencylace cabled data networks
in,
phasis added by Maguire]
Maguire Wireless [email protected] 2006.03.12
Wireless LANs“The wireless workplace will soon be upon us1
Telia has strengthened its position within the area of
radio-baacquisition of Global Cast Internetworking. The
companyMobile’s offering in wireless LANs and develop solutions
thathe wireless office. A number of different alternatives to
fixedunder development and, later wireless IP telephony will
also
…
The acquisition means that Telia Mobile has secured the
rescontinued expansion and product development within the
fieldRadio LANs are particularly suitable for use by small and
moperators of public buildings such as airports and railway sta
Today’s radio-LAN technology is based on inexpensive
procertification. They are easy to installand are often used to
repfor example, large buildings.
…” [em
1. Telia press annoucement: 1999-01-25
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orporate network from airports, centers, etc. via WLAN.
Maguire Telia’s [email protected] 2006.03.12
Telia’s HomeRunhttp://www.homerun.telia.com/
A subscription based service to link you to your ctrain
stations, ferry terminals, hotels, conference
Look for Telia’s HomeRun logo:
http://www.homerun.telia.com/
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Module 1: 57 of 84Practical Voice Over IP (VoIP): SIP and
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Net"rated GSM- IP (Internet
applicationsea GSM coverage to
lications includeling, unified messagingharing using voice
Maguire Ericsson’s "GSM on the Net"[email protected]
2006.03.12
Ericsson’s "GSM on the• Provide communication services over an
integ
Protocol) network• support local and global mobility• support
multimedia capabilities and IP-based • uses small radio base
stations to add local-ar
office LANs• provides computer-telephony integration: app
web-initiated telephony, directory-assisted diaand advanced
conferencing and application-sdatacoms and video.
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phonyfrom the US is via VoIP, basedia PSTN [10] (the article
citesGroup/Primetrica Inc.)
plans for unlimited North
ce their exchanges with IPn, “Telia ersätter all AXE med05, page
4.
e [11].
Maguire VOIP vs. traditional [email protected]
2006.03.12
VOIP vs. traditional teleAs of 2003 approx. 14% of International
traffic to/on 24 billions minutes vs. 170.7 billion minutes vthe
source of data as TeleGeography Research
As of December 2004, commercial VoIP callingAmerican traffic
cost ~US$20-30/month.
There is a move for traditional operators to replatelephony, see
Niels Herbert and Göte AnderssoIP-telefoni”, Elektronik Tidningen,
#3, 4 March 20
For information about the development of the AXE switches se
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y Bart Stuck and Michaelolume 28, Number 8,August
ephony, and what is hype?
at in 1998, access arbitrage isticipate that
switched-accessisappears and/or access rates
d data via packetized networkscosts. As a result, VOIP willd
voice. Indeed, as voice/data
voice becoming economically
ion means that ISPs do not pay the ISP justreceives calls
from
Maguire [email protected] 2006.03.12
Economics“Can Carriers Make Money On IP Telephony?” bWeingarten,
Business Communication Review, V1998, pp. 39-44.
"What is the reality in the battle over packet-versus-circuit
tel
Looking at the potential savings by cost element, it is clear
ththe major economic driver behind VOIP. By 2003, we anarbitrage
will diminish in importance, as the ESP exemption ddrop to true
underlying cost.
However, we believe that the convergence between voice anwill
offset the disappearance of a gap in switched accesscontinue to
enjoy a substantial advantage over circuit-switcheconvergence
occurs, we see standalone circuit-switchednonviable."
Note: Enhanced Service Provider (ESP) exemptaccess charges to
local phone companies {sinceusers}
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honyuld carriers worry?”1 nicely
re ⇒ Content-neutralhe large margins which
d landline): $1.70/MB”
can offer phone services traffic
s2
can create a service
traditional telephony services.
uate Course "Internet Multimedia", University of Oulu, 3-6
Maguire VoIP vs. traditional [email protected]
2006.03.12
VoIP vs. traditional telepHenning Schulzrinne in a slide
entitled “Why shostates the threats to traditional operators:
• Evolution from application-specific infrastructubandwidth
delivery mechanism - takes away tthe operators are used to (and
want !):
– “GPRS: $4-10/MB, SMS: >$62.50/MB, voice (mobile an
• Only operators can offer services ⇒ Anybody• SIP only needs to
handle signaling, not media
• High barriers to entry ⇒ No regulatory hurdleIn addition to
this we can add:
• Only vendors can create services ⇒ anybodyNB. These new
services can be far broader than
1. Henning Schulzrinne, “When will the telephone network
disappear?”, as part of Intensive GradJune 2002.
2. see “Regulations in Sweden” on page 76
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t least this patent:
formation among a plurality ofrotocol contemplate first ands are
coupled to both the firstpath and writes another signalhich
electrically precedes thensmitted in a regular, cycliccycle code
for enabling eachket to transmit, it can read theo, a logical
interpretation may
oice and data transmission
Maguire [email protected] 2006.03.12
PatentsMixing voice and data in the LAN goes back to a
ABSTRACT: In order to control the transfer of packets of
instations, the instant communications system, station and psecond
oppositely directed signal paths. At least two stationand the
second signal paths. A station reads one signal from aon the path.
The one signal is read by an arrangement warrangement for writing
the other signal. Packets are trasequence. A head station on a
forward path writes a startstation to transmit one or more packets.
If a station has a pacbus field of a packet on the forward path.
Responsive theret
US 4581735 : Local area network packet protocol for combined
v
INVENTORS: Lois E. Flamm and John O. Limb
ASSIGNEES: AT&T Bell Laboratories, Murray Hill, NJ
ISSUED: Apr. 8 , 1986
FILED: May 31,1983
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sy. If the path is not busy, theereon including the busy
field.ath is detected as not busy. Ination may write different
startble stations to transmit voicemit data packets, etc. for
the
be written in a regular, e.g.,h clipping. Still further, the
lastackets on a reverse path forResponsive to the control
the respective stations to, fore number of packet time
slots,
Maguire [email protected] 2006.03.12
be made as to whether the forward path is busy or is not
bupacket may be written on the path by overwriting any signal thIf
the path is busy, the station may defer the writing until the
porder to accommodate different types of traffic, the head stcycle
codes. For example, a start-of-voice code may enapackets; a
start-of-data code may enable stations to transdifferent types of
traffic. Further, the start cycle codes mayperiodic, fashion to
mitigate deleterious effects, such as speecstation on the forward
path may write end cycle codes in pcommunicating control
information to the head station.information, the head station may
modify the cycle to permitexample, transmit more than one packet
per cycle or to vary thwhich are allocated to each of the different
types of traffic.
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nds
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IPing voice-over-frame relay service in 40ending voice traffic
over its frame relayHuntley, executive VP of marketing for
c over data networks. Most of the major
US$3-4/month US$50-60/month for unlimiteds not include equipment
attion} The Service Level5ms round trip latency, and
e US for US$34.95/monthe future will be about thehere these two
worlds meet is
y,1/98.
Maguire Carriers offering [email protected] 2006.03.12
Carriers offering VO“Equant, a network services provider, will
announce tomorrow that it is introduccountries, ... The company
says customers can save 20% to 40% or more by snetwork. "This is
the nearest you’re going to get to free voice," says LaurenceEquant
Network Service. … Equant isn’t alone in its pursuit to send voice
traffi
carriers are testing services that would send voice over data
networks. ... .”1
• October 2002:• Verizon offering managed IP telephony via IPT
Watch for • WorldCom offering SIP based VoIP for DSL customers
for
local, domestic long distance, and data support {price
doeUS$200-300 per phone and DSL/Frame relay/ATM connecAgreement
(SLA) specifies >99.9% network availability, 99.5% packet
delivery.
• December 2004:• Verizon offering VoiceWing - with unlimited
calling within th• “As we see the industry fundamentals continue to
shift, th
convergence of computing and telecommunications. And wwhere MCI
will be.” -- Michael D. Capellas, MCI CEO 2
1. Mary E. Thyfault, Equant To Roll Out Voice-Over-Frame Relay
Service, InformationWeek Dail10/2
2. http://global.mci.com/about/publicpolicy/voip/
http://global.mci.com/about/publicpolicy/voip/
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onnection
its own staff!)
lly 1+ move/person/year)
yond the LAN to the WANtilize a Web browser to
nt, thus limiting
uipment (CPE)
Maguire MCI (formerly WorldCom) [email protected]
2006.03.12
MCI (formerly WorldCom) CPreviously
• 3 or more separate networks (often each had •
Duration/geography-based pricing
• Expensive moves, adds, and changes (typica• Standalone
applications - generally expensive• Closed PBX
architectureToday
• via gateway to the PSTN, service expands be• centralized
intelligence is offered; customers u
control and manage their network• MCI incurs the costs of buying
major equipme
customer’s risk and capital investment• One source for all
services• Easy mobility• Choice of vendors for Customer Premises
Eq
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s Inc.alling service across the Unitedoice over Internet
protocol
Offerings, Internetweek.com,January 13, 2004,
ticle.jhtml?articleID=17300739
VoIP and data services to their
voice signals from the PSTNoice signals when a call is
routed
of calls per month - as of
Maguire Level 3 Communications [email protected]
2006.03.12
Level 3 CommunicationIntroduced (3)VoIP Toll Free service: “a
toll-free cStates, rounding out its local and long distance
vofferings.”
Antone Gonsalves, E-BUSINESS: Level 3 Rounds Out VoIP
http://www.internetweek.com/e-business/showAr
Level 3 sells services to carriers, who then offer
customers.
Usessoftswitch networking technology to convertto IP packets and
conversely converts packets to v
to the public switched network. (>30 x 109 minutes January
13, 2005)
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elefoniential broadband telephonytsip AB (www.hotsip.com ). In
calls, presence, and instant
st 80 kr. price as if you calledork.area/city” code 075
(i.e.,
00-numbers)
Maguire TeliaSonera [email protected]
2006.03.12
TeliaSonera BredbandstFebruary 5th, 2004 TeliaSonera annouces
theirresidservice using server and client products from Hoaddition
to telephony, the service includes: videomessaging.[6]
• The startup cost is 250 kr and the monthly co• Calls to the
fixed PSTN network are the same
from a fixed telephone in their traditional netw• Customers get
a telephone number from the “
+46 75-15xxxxxxx)• They do not support calls to “betalsamtal”
(09
www.hotsip.com
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Nff” when it can really emulateSTN:
strings}
Maguire Emulating the [email protected] 2006.03.12
Emulating the PSTMany people feel that VoIP will really only
“take oall the functions which users are used to in the P
• Integration with the web via: Click-to-connect• “Dialing” an
e-mail address or URL {digits vs. • Intelligent network (IN)
services:
• Call forward, busy• Call forward, no ans.• Call forward,
uncond.• Call hold• Call park• Call pick-up• Call waiting•
Consultation hold• Do not disturb• Find-me• Incoming call
screen/Outgoing call screen• Secondary number in/Secondary number
out• Three-way conference• Unattended transfer
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ans providing functionstil 31 August 2005”)ng Desktop calles,
etc.asing use of Power over the wall outlet does not
location of the station?}
Maguire Emulating the [email protected] 2006.03.12
• additional PBX features (which in Sweden mesuch as “I’m on
vacation and will not return un
• Computer-Telephony Integration (CTI), includimanagement,
integration with various databas
• PSTN availability and reliability (thus the increEthernet for
ethernet attached IP phones - sohave to provide power for the phone
to work)
• Roaming - both personal and device mobility• Phone number
portability• E911 service {How do you handle geographic
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turesl
ld be the target of a call
Maguire Calling and Called [email protected]
2006.03.12
Calling and Called Fea• Calling feature - activated when placing
a cal
• e.g., Call Blocking and Call Return
• Called feature - activated when this entity wou• Call
Screening and Call Forward
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& Instant
traveling, …
ce, video, …
nce and Instant Communications), Protocols, and
Applications.
ssaging platform for thers each week.
oyees - an experimentalging (IM), email, voice,
Maguire Beyond the PSTN: Presence & Instant
[email protected] 2006.03.12
Beyond the PSTN: PresenceMessaging
• Presence , i.e., Who is available?• Location , i.e., Where are
they?: office, home,• Call state : Are they busy (in a call) or
not?• Willingness : Are they available or not?• Preferred medium :
text message, e-mail, voi• Preferences (caller and callee
preferences)
See Sinnreich and Johnston’s Chapter 11 (Prese& course2G5565
Mobile Presence: Architectures
• Reuters has deployed a SIP-based instant-mefinancial services
industry that has 50,000 use
• IBM’s NotesBuddy application for ~315k emplmessaging client
that integrates instant messaand other communication.
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vices
onal time”
roup chat sessions, …f people are all available
his is a good area fortures, Protocols, and
ensions (SIMPLE)
Maguire Presence-Enabled [email protected]
2006.03.12
Presence-Enabled Ser• Complex call screening
• Location-based: home vs. work• Caller-based: personal friend
or business colleague• Time-based: during my “working hours” or
during my “pers
• Join an existing call ⇒ Instant Conferencing, g• Creating a
conference when a specific group o
and willing to be called• New services that have yet to be
invented! (T
projects in 2G5565 Mobile Presence: ArchitecApplications)
• SIP Messaging and Presence Leveraging ExtWorking Group was
formed in March
2001http://www.ietf.org/html.charters/simple-charter.html
http://www.ietf.org/html.charters/simple-charter.html
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for VoIP
handsets controlled by thees onany platform
edia contentin digital form
s
Implementation
H.323
Softswitches
ession Initiation Protocol (SIP)
Maguire Three major alternatives for [email protected]
2006.03.12
Three major alternatives
SIP⇒ a change from telephony’s “calls” betweennetwork to
“sessions” which can be betweenprocessanywherein the Internet and
with bothcontrol andmand hence can be easily manipulated.
• thus a separate voice network is not necessary• open and
distributed nature enables lots of innovation
– since both control and media can be manipulated and– “events”
are no longer restricted to start and end of call
Concept
Usesignalling concepts from the traditional telephony
industry
Usecontrol concepts from the tr