Questions about codecs

[Guest]

Archived from groups: comp.dcom.voice-over-ip (More info?)

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: http://www.uninett.no/voip/codec.html
but it doesn't help a lot

 

[Guest]

Archived from groups: comp.dcom.voice-over-ip (More info?)

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






"Mark De Biasi" <nowhere@null.to> wrote in message
news:3697l7F4vhvi5U1@individual.net...
> 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: http://www.uninett.no/voip/codec.html
> but it doesn't help a lot

 

[Guest]

Archived from groups: comp.dcom.voice-over-ip (More info?)

Better correct this b4 someone flames me!!!

> Less data through put = More bandwidth<

Less data through put = More bandwidth AVAILABLE

HubSwitch
A slip of the KB



"HubSwitch" <HubSwitch@routers.hub.switch> wrote in message
news:ctnqmo$nlh$1@newsg2.svr.pol.co.uk...
> 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
>
>
>
>
>
>
> "Mark De Biasi" <nowhere@null.to> wrote in message
> news:3697l7F4vhvi5U1@individual.net...
> > 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: http://www.uninett.no/voip/codec.html
> > but it doesn't help a lot
>
>

 

[Guest]

Archived from groups: comp.dcom.voice-over-ip (More info?)

HubSwitch wrote:
> Better correct this b4 someone flames me!!!
>
>
>>Less data through put = More bandwidth<
>
>
> Less data through put = More bandwidth AVAILABLE

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?

 

[Guest]

Archived from groups: comp.dcom.voice-over-ip (More info?)

Mark De Biasi wrote:

> 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?

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.

> 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: http://www.uninett.no/voip/codec.html
> but it doesn't help a lot

 

[Guest]

Archived from groups: comp.dcom.voice-over-ip (More info?)

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)




"Mark De Biasi" <nowhere@null.to> wrote in message
news:369domF4s2r2tU1@individual.net...
> HubSwitch wrote:
> > Better correct this b4 someone flames me!!!
> >
> >
> >>Less data through put = More bandwidth<
> >
> >
> > Less data through put = More bandwidth AVAILABLE
>
> 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?

 

[Guest]

Archived from groups: comp.dcom.voice-over-ip (More info?)

"Mark De Biasi" <nowhere@null.to> wrote in message
news:3697l7F4vhvi5U1@individual.net...
> 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?

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.

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

Actually we do: it's iLBC (www.google.com.hk/search?q=ilbc+skype). 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.

> 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?

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

> Which one is the best?

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
(http://www.globalipsound.com/legal/licenses.php). iLBC has a fixed 13.3
Kbit/s bitrate, whereas Speex is multi-rate (see
http://www.speex.org/comparison.html ).

> Which one is the most used? (I assume that both phones must use the same
> one when talking to each other)

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.

> so far, I found this page: http://www.uninett.no/voip/codec.html
> but it doesn't help a lot

This one may give you a good background:

http://www.vocal.com/data_sheets/audio_codecs.html

Enzo

 

[M]

Archived from groups: comp.dcom.voice-over-ip (More info?)

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:
> 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: http://www.uninett.no/voip/codec.html
> but it doesn't help a lot