Questions about codecs

Am new to this too, the hyperlink on your email helps me.

Less data through put = More bandwidth "Zipping, (CODEC) " files to make big packages small = good- with cpu trade off = Latency

More expence = Less bandwidth taken, speech quality and less Latency... Says it all really LESS expence =More bandwidth taken, lower speech quality and possibly more latency (due to larger data chunks to process)

All explanations are Opinions from HubSwitch (my get out clause) As i said =new to this and could get it wrong :S, but I am confident........ 4 now :)

Reply to
HubSwitch
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Better correct this b4 someone flames me!!!

Less data through put = More bandwidth AVAILABLE

HubSwitch A slip of the KB

Reply to
HubSwitch

Hi,

I am experimenting VOIP for the first time. Although I have some IT experience, this is a different field, and I get sometimes confused when I read the web sites that illustrate it.

To start, I have tried Skype, and it works rather well. Then I tried babble.net with X-lite, and it rather works well too. I had tried X-lite with SIPphone from PC to PC (both in europe with broadband), and it didn't work.

Does the quality of the phone call (in the user sense of the expression) depend, among other things, from the codec used?

I know that with Skype there is one proprietary codec, and we don't know.

But with SIP, X-lite, I see: g711u and g711a (disabled), and gsm, iLBC and Speex (enabled).

Which enabled codec is actually used when I talk? Which one is the best? Which one is the most used? (I assume that both phones must use the same one when talking to each other)

so far, I found this page:

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it doesn't help a lot

Reply to
Mark De Biasi

which is good if you don't have a lot of bandwidth available.

but, in terms of the quality of the sound and of packet loss recuperation, which one is advisable?

Reply to
Mark De Biasi

ok, I found the answer to this: it is GSM (with babble.net), and you can see it because it get sorrounded by a square on the phone display.

Reply to
Uno Qualunque (fu: Fil)

Now your really splitting hairs:

The human voice can be recorded at very low levels (sampling rates) and still be understood with ease, the higher the sample rate the easyier it is to listen to:

AM Radio compared to a CD recording in DDD theres nooooo comparison!

Or Telephone converstaion compared to a FM (DAB) Broadcast.....

Though... do you NEED to have full surroud sound 5.1 DTS listening concept just to listen to a monophonic voice peice thats saying "i'll be late for dinner hunny" or "Hello Dad/Mom/Son/Daughter"

HubSwitch

PS To answer your question, the CODEC that has the higher DATA (sample rate) and least LATENCY would be the one to use in my opinion! (look at the figures to work that one out to use with your ADSL/56k modem)

Reply to
HubSwitch

Yes, also with regard to latency. Unfortunately, achieving at the same time a high data compression rate and good sound quality requires both a smart algorithm (often patented) and lots of CPU cycles.

Actually we do: it's iLBC

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. This doesn't help Skype to interoperate with standards-based VoIP systems (either SIP or H.323) because there are many other proprietary (and undocumented) parts in its protocol stack.

With SIP (or, more precisely, SDP), the codec is negotiated between the two sides resulting in the choice of the first in the list supported by both. In X-lite

Well, it depends. G711 has no compression: it uses sequences of bytes representing PCM samples taken 8000 times a second. In order to use only 8 bits but keep a 12-bit resolution of small signals, the encoding of each sample is near-logarithmic; the u and A versions just use a different conversion table. Here you get low latency, good fidelity, minimum CPU load but high bit rate (64 Kbit/s).

GSM, or more accurately the "full rate" GSM 06.10, is a codec used several years ago on GSM cellular phones; modern handsets and cellular providers replaced it with two others, the "half rate" GSM 06.20 and the "enhanced full rate" GSM 06.60. The reason for GSM 06.10's popularity is that it's relatively unencumbered by patents (despite some claims from Philips) and there is a much used opensource implementation by Jutta Degener and Carsten Bormann. GSM uses 13 Kbit/s (almost five times less than G711) and is not too demanding on the CPU side. Unfortunately the quality, although acceptable, is not extremely good - which is why in GSM networks it was superseded by GSM 06.60...)

iLBC and Speex are both quite good and unpatented, although iLBC comes with some strings attached

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iLBC has a fixed 13.3 Kbit/s bitrate, whereas Speex is multi-rate (see
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).

Most commercial products support G.711a/u, and very few, if any, GSM

06.10, iLBC or Speex. For low bitrates they tend to use proprietary codecs available on commercial basis, mostly G.729 with SIP and G.723.1 with H.323.

In opensource projects, Speex has become quite popular displacing GSM

06.10, and there is an onging effort to standardise its use in RTP payload.

This one may give you a good background:

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Enzo

Reply to
Enzo Michelangeli

gsm, iLBC GSm is like low quality but smaller packets .. now ilbc is got to be the best for quality and slower broadband connections.. people rate it to the g729a ..(requires a licence)

anyway its free (iLBC) and you want that enabled and ulaw 711 and gsm the rest i dunno about and take alot of bandwidth.. 711 takes quite a bit compared to gsm and iLBC

hope that helps.. with out the mumbo jumbo..

m.

Mark De Biasi wrote:

Reply to
m

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