IP Telephony : Packet Based Multimedia Communications Systems

by Hersent, Olivier

• ISBN13: 9780201619102
• ISBN10: 0201619105
• Format: Paperback
• Copyright: 2000-01-01
• Publisher: Addison-Wesley Professional

Summary

This book provides a comprehensive practical overview of the technology behind Internet Telephony, giving essential information to IT professionals who need to understand the background and explore the issues involved in migrating the existing telephony infrastructure to an IP based real time communication service. Assuming a working knowledge of IP and ISDN networking, it addresses the technical aspects of real-time applications over IP, with an in-depth coverage of voice and video applications and protocols. Drawing on their extensive research and practical development experience in VoIP from its earliest stages, the authors give you access to all the relevant standards and cutting-edge techniques in a single resource.

Table of Contents

Foreword	xi
Jeff Pulver
Preface	xiii
SECTION 1 The application layer, IP telephony protocols	1	(194)
H.323 and a general background on IP telephony	3	(118)
A little history	4	(2)
Where to find documentation	4	(1)
From RTP to H.323: a quick tour	5	(1)
Transporting voice over a packet network	6	(14)
A Darwinian view of voice transport	6	(3)
Voice and video over IP with RTP and RTCP	9	(11)
H.323 step by step	20	(22)
The `hello world' case: simple voice call from terminal A to terminal B	21	(7)
A more complex case: calling a public phone from the Internet	28	(7)
H.323 across multiple domains	35	(7)
Advanced topics	42	(35)
Faster procedures	42	(8)
Conferencing with H.323	50	(6)
Directories and numbering	56	(8)
H.323 security: H.235	64	(13)
Media streams	77	(33)
Codecs	77	(23)
DTMF	100	(2)
Fax	102	(8)
Supplementary services using H.450	110	(8)
H.450.1	111	(1)
H.450.2: call transfer	111	(5)
H.450.3: call diversion	116	(2)
The future of H.450	118	(3)
Future work on H.323	118	(3)
The session initiation protocol (SIP)	121	(42)
The origin and purpose of SIP	122	(14)
Overview of a simple SIP call	122	(6)
SIP messages	128	(7)
Session description syntax, SDP	135	(1)
Advanced services with SIP	136	(18)
SIP entities	136	(8)
User location and mobility	144	(5)
Multiparty conferencing	149	(3)
Configuring network-based call handling	152	(1)
Billing SIP calls	152	(2)
SIP security	154	(3)
Media security	154	(2)
SIP firewalls	156	(1)
SIP and H.323	157	(3)
What SIP does and H.323 does not	157	(1)
What H.323 does and SIP does not	158	(2)
H.323 to SIP gateways	160	(1)
Conclusion on the future of SIP and its relation to H.323	160	(3)
Media gateway to media controller protocols (MGCP)	163	(32)
Introduction	164	(12)
Which protocol?	165	(1)
What about the requirements?	166	(1)
A new architecture for IP telephony?	167	(9)
What is MGCP?	176	(5)
MGCP commands	177	(4)
Protocols at work	181	(14)
Scenario 1	181	(4)
Scenario 2	185	(5)
The H.323 case	190	(5)
SECTION 2 Voice technology background	195	(98)
Voice quality	197	(32)
Introduction	198	(2)
Reference connection	198	(2)
Echo in a telephone network	200	(9)
Talker echo, listener echo	200	(1)
Hybrid echo	201	(3)
Acoustic echo	204	(1)
How to limit echo	205	(4)
Delay in a telephone network	209	(11)
Influence of the operating system	209	(1)
Influence of the jitter buffer policy on delay	210	(1)
Influence of the codec, frame grouping and redundancy	211	(3)
Consequence for measuring end-to-end delay	214	(2)
Acceptability of a phone call with echo and delay	216	(3)
Consequences for an IP telephony network	219	(1)
The approach of ETSI TIPHON	220	(9)
Other requirements for gateways and transcoding equipment	227	(2)
Voice coding	229	(64)
Introduction	230	(17)
Transmitted bandwidth, sampling and quantization	230	(4)
Some basic tools for digital signal processing	234	(3)
The A or u law ITU-T 64 kbit/s G.711	237	(3)
Specifications and subjective quality of speech coders	240	(3)
ACR subjective test or what is the MOS	243	(4)
Speech and auditory properties	247	(6)
Speech production	247	(4)
Auditory perception used for speech and audio bitrate reduction	251	(2)
Quantization and coders	253	(17)
Adaptive quantizers	253	(3)
Differential (and predictive) quantization	256	(5)
Vector quantization	261	(1)
Entropy coding	262	(1)
Waveform coders: the ADPCM ITU-T G.726	262	(6)
Wideband speech coding using waveform-type coder	268	(2)
Speech coding techniques	270	(18)
Hybrids and analysis by synthesis speech coders	270	(2)
The GSM Full Rate RPE-LTP speech coder	272	(2)
Codebook excited linear predictive (CELP) coders	274	(5)
The ITU-T 8 kbit/s CS-ACELP G.729	279	(2)
The ITU-T G.723.1	281	(2)
Discontinuous transmission and comfort noise generation	283	(1)
The low delay CELP coding scheme: ITU-T G.728	284	(3)
Partial conclusion on speech coding techniques and their near future	287	(1)
Remarks applicable to VolP telephone gateways	288	(5)
The electrical echo canceller	289	(1)
Best effort	289	(1)
Non-standardization	290	(3)
SECTION 3 The network	293	(128)
Quality of service	295	(46)
What is QoS?	296	(45)
Describing a stream	297	(1)
Queuing techniques for QoS	298	(9)
Signaling QoS requirements	307	(9)
RSVP	316	(7)
Scaling issues with RSVP	323	(3)
Classes of service in the backbone	326	(4)
RSVP to Diffserv mapping	330	(3)
Improving QoS in the best-effort class	333	(4)
Issues with slow links	337	(3)
Conclusion	340	(1)
Network dimensioning	341	(36)
Simple compressed voice flow model	342	(12)
Voice coders	342	(2)
Model for N simultaneous conversations using the same coder	344	(2)
Loss rate and dimensioning	346	(6)
Packet or frame loss?	352	(1)
Multiple coders	353	(1)
Network dedicated to IP telephony	354	(4)
Is it necessary?	354	(1)
Network dimensioning	354	(4)
Merging data communications and voice communications on a common IP backbone	358	(3)
Prioritization of voice flows	358	(3)
Impact on end-to-end delay	361	(1)
Multipoint communications	361	(7)
Audio multipoint conferences	361	(6)
Multipoint video conferencing	367	(1)
Modeling call seizures	368	(6)
Introduction to the Erlang model	368	(2)
Model for a limited set of servers and calls are rejected if no server is available	370	(1)
Calls per second	371	(3)
Conclusion	374	(3)
Multicast routing	377	(44)
Introduction	378	(1)
When to use multicast routing	378	(3)
A real-time technology	378	(1)
Network efficiency	379	(2)
The multicast framework	381	(5)
Multicast address, multicast group	381	(1)
Multicast on Ethernet	382	(1)
Group membership protocol	383	(3)
Controlling scope in multicast applications	386	(3)
Scope versus initial TTL	386	(1)
TTL threshold	387	(1)
Administrative scoping	387	(2)
Building the multicast delivery tree	389	(4)
Flooding and spanning tree	389	(1)
Shared trees	389	(1)
Source-based trees	390	(3)
Multicast routing protocols	393	(9)
DVMRPv3	393	(4)
Other protocols	397	(4)
Core-based trees	401	(1)
Security issues in IP multicast	402	(2)
Unauthorized listening	402	(1)
Unauthorized sending and denial of service attacks	402	(1)
Firewalls	403	(1)
The MBONE	404	(2)
Routing protocols	404	(2)
Inter-domain multicast routing	406	(5)
Interoperational between domains running different protocols	406	(2)
BGMP	408	(3)
Conclusion on multicast inter-domain routing	411	(1)
Multicast caveats	411	(3)
Multicasting on non-broadcast media	411	(2)
Flooding	413	(1)
Common issues	413	(1)
Address allocation	414	(2)
MBONE applications	416	(5)
Video conferencing with RTP on multicast networks	416	(1)
SDR: session directory	417	(1)
VIC and VAT	417	(1)
Reliable multicast	417	(4)
Appendix: well known multicast addresses	421	(6)
References	427	(12)
Glossary	439	(12)
Index	451

Excerpts

This book is about the newest service to arrive on the Internet: interactive voice and video. Since its first appearance to the market in 1995, it is amazing to note how fast the IP telephony technology has evolved. But this evolution alone wouldn't account for VoIP's growing success if there weren't a general recognition of the importance of the Internet in our day-to-day activities. Flashback In 1991, the Internet was still a maze without a map where only addicts really felt at home. As students we used to print and exchange our bookmarks - as soon as someone had found a really good ftp site with anonymous access, he would copy and distribute the listing. Any good book on the Internet inevitably ended with hundreds of pages of such bookmarks. The Internet was a nice tool, especially email and chat, but really you had to enjoy being stuck in front of your black and white computer screen for hours. Then, in 1993, with the World Wide Web, came the first revolution. The Internet became colorful and our piles of paper bookmarks were instantly obsolete, replaced by links on web pages. Soon there were too many links for one person to keep track of. So two US students came up with the idea of keeping bookmarks up to date for other people and created Yahoo!. It was a good thing indeed. Thus, with all the bookmark sites and search engines, the Internet became less intimidating for non-UNIX gurus. It became feasible for a literature student to get an Internet account. By 1996 every student had an email address and most of them actually used it. But for most people, the Internet was still a toy. It was too complex to reach the mass market, and would never really count outside universities and research labs. In one sentence: 'the Internet will really count when your grandmother can use it.' It was around that time that we saw the first attempts to build an Internet telephony gateway. The first prototype was a strange hybrid. The telephone interface was a modem with speakerphone capabilities. The problem with this speakerphone modem was that there was no way to simultaneously play a sound and record from a PC. Actually you could use the modem only to dial the destination number. Some sound boards had a driver that made it possible to simultaneously play and record ('full-duplex'), but no telephone interface, so you had to wire the sound-board line-in jack to the modem microphone, and the modem speaker to the sound-board line-out jack. Of course, some software was needed to turn this into an Internet telephony gateway, but in 1996 there was already good Internet telephony freeware available, such as VAT, and adding some code to interface with the modem wasn't too difficult: when an incoming call arrives, pick up the line, play a welcome prompt, get the destination number with DTMF touchtones (the modem had that capability), relay the information to the destination gateway that would use the modem to dial the right number, and spawn the Internet telephony software. This was only a very crude, one-line gateway, but its potential was immense. The telephone network is the example of a technology that really counts. There are over 800 million telephone lines in the world, and a five-year-old can use a telephone. Obviously, being able to carry even a small fraction of all telephone conversations over the Internet would have been a major achievement. Many research labs suddenly realized that it could be a second Internet revolution and tried to evaluate if it was possible to build a more sophisticated gateway. This was only three years ago, but already today, there are more and more companies running their operations without the use of a single 'blackphone' of a PBX. And tomorrow, our grandmother may still use a regular analogue telephone, but