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Getting started in IP telephony

I've started work for a small company in the UK installing and configuring IP telephony systems - in particular the Avaya IP office and argent office range. I've done lots of software development (mainly C under Windows) but I'm not knowledgeable about IP telephony or larger network environments (all this was discussed in detail before I started and the company and I are prepared for a learning curve).

Is there a good introduction to IP telephony that gives a high level overview? I've got all the manuals and trial equipment to play with, but this is in-depth info and I'm still getting my head round it all.

Mr Avaya
Wednesday, November 05, 2003

There are lots of on-line resources. Most academic work is based on SIP. Commercial stuff covers H.323, MEGACO and SIP.

If you're into SIP, you should check out www.vovida.org. They provide an open source VoIP infrastructure aimed at linux. They also have some tutorials.

http://www.vovida.org/document/Training/1_VoIP_Overview/index_files/v3_document.htm

The main difference between H.323 and SIP is that H.323 tries to cover all aspects of VoIP, while SIP only covers the signalling. For transport, SIP could use anything, but in practice it is almost always RTP.

Actually, VoIP is a misnomer, since SIP, and in some degree H.323, is not only about voice. It's about interactive multimedia conversations.

I'm more familiar with SIP, so I'll just drop some references here.

http://www.sipcenter.com/aboutsip/whatissip.html
http://www.ietf.org/html.charters/sip-charter.html
http://www.cs.columbia.edu/sip/

The IETF SIP working group is currently very busy. People are  proposing extensions left and right. But at is't core, SIP is still rather simple.

Hope it helps

henrik
Wednesday, November 05, 2003

Henrik - thanks for the info about SIP. I also have to deal with telephony over Cat5 data cabling (I think it's called Convergence) and mixed installations where there is original telco wiring, mixed with newer cat5 cabling. In fact, using the data cabling to piggy back the phone system without using VoIP is most of what I will be doing.

Also this job is unlikely to involve coding or low level work. The Avaya IP office handles most of the configuration. I just need to understand where it fits in the telephony world.

Mr Avaya
Wednesday, November 05, 2003

Unless there is CSMA/CD packet data, e.g ethernet frames, in the cabling, you should not need to set up any complicated resource reservation schemes. If you have to do that, it can get really ugly....

I suppose you have some product to deal with the link layer differences between standard telco cabling and data cabling. If not I forsee some low level coding ahead.

Good luck, and have fun :)

henrik
Wednesday, November 05, 2003

Mr Avaya,

If you want to email a question to me, I can try to answer it. I used to be a VoIP engineer and designed a VoIP product once. My background is in voice and data.

Basically, voice gets encoded into a 64kbps channel (8-bit samples * 8kHz) and transmitted over twisted pair copper wire. This DS0 (digital signal zero) channel is then aggregated into a DS1 signal (24 DS0s) and transmitted as a T1. This T1 is usually called a trunk and goes to a multiplexer, the central office, or wherever your calls are processed. Since you are in the UK, this is called an E1 and has 32 voice channels instead.

When you start getting into IP over Ethernet, you can either transmit this stuff over the telco copper pair using a different frequency band or encode the voice stuff into packets/datagrams and transmit it over Cat5 or fiber. Once you start sending it over Ethernet, you will probably encode it using common protocols such as Ethernet (L1/2), IP (L3), UDP (L4), and RTP (also L4). There are many ways to do call control and these include SIP and H.323 among others.

In enterprise settings, perhaps like yours, you will probably configure only a few setting such as the IP addresses of your media and signaling gateways and the codec you will use for voice encoding. G.711 is the most common, but G.728 and G.729 are also popular because they offer higher compression. Latency (packet delay) and jitter (packet sequencing) must also be engineered properly. Echo cancellation is also important as is security and quality of service.

My problem with VoIP is that to maintain quality of service, you either have to have a lot of bandwidth or start playing games with RSVP or MPLS to reserve bandwidth and set priorities. When you consider the protocol overhead of Ethernet, IP, UDP, RTP, etc., you are basically using more bandwidth than you were with old-fashioned 64kbps voice.

Oh, and some systems use encryption as well for security. This adds additional overhead, especially in the computational power of the network elements you're using.

StickyWicket
Wednesday, November 05, 2003

StickyWicket

Many thanks - this is exactly what I need. I have reasonable understanding of IT architecture, but when I read the Avaya product info there are lots of acronyms and concepts I'm unfamiliar with.

Eg I understand the ethernet principle whereby packets flow down the wire, but I'm still puzzled how telephony can use the same cable for voice (I *don't* mean VoIP in this case, what I mean is how does telephony use part of the same cat5 cable? by using a different frequency band?) And lots of similar questions.

As you said, I don't need to configure or setup any of this - Avaya IP Office handles everything at a higher level. Eg it lets you define extensions, phone groups, forwarding, voicemail etc. But I need to know in general terms what is going on, to do my job properly.

So what I am looking for, is something like an O'Reillys 'In a Nutshell' book that gives an overview in 200 or 300 pages of what is happening.

Mr Avaya
Wednesday, November 05, 2003

Most networking encyclopedias cover what you want to know. LAN Times and Newton's are both good. Basically, a copper twisted pair can handle a wide spectrum of frequencies. Ethernet would use some of this spectrum (band) and voice could use another band. This is basically how DSL works. Your normal voice occupies the lower end of the spectrum as normal, and the data is modulated over a higher, and much larger band. A filter at the central office splits the high frequency stuff off and converts it back into IP datagrams. Since high frequencies get attentuated easily, this is why DSL users must be within a certain distance of the central office (or access device). Otherwise, the signal would be lost.

StickyWicket
Wednesday, November 05, 2003

Is there any useful, intermediate reference that gives an overview of the field? I have a good knowledge of programming and some electronics knowledge (I used to be a radio amateur). The only useful book I have is "Data Communications, Open Networks and Open Systems" by Fred Halsall. This is somewhat out of date and concentrates on the network side, not the telephony.

Also I've discovered the topic should be 'Network telephony' (?) not 'IP Telephony' since voice does not have to be sent using IP packets. BTW thanks for your comments - they are very helpful.

Mr Avaya
Thursday, November 06, 2003

Hey, if you want to hack it there's always the fact that not every pair in a cat5 cable is used, just take an unused pair and put a phone jack on it!

/neophyte

Squeebee
Friday, November 07, 2003

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