Questions about codecs

Discussion in 'VOIP' started by Mark De Biasi, Feb 1, 2005.

  1. 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
    Mark De Biasi, Feb 1, 2005
    #1
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  2. Mark De Biasi

    HubSwitch Guest

    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" <> wrote in message
    news:...
    > 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
    HubSwitch, Feb 1, 2005
    #2
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  3. Mark De Biasi

    HubSwitch Guest

    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" <> wrote in message
    news:ctnqmo$nlh$...
    > 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" <> wrote in message
    > news:...
    > > 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

    >
    >
    HubSwitch, Feb 1, 2005
    #3
  4. "Mark De Biasi" <> wrote in message
    news:...
    > 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
    Enzo Michelangeli, Feb 1, 2005
    #4
  5. 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?
    Mark De Biasi, Feb 1, 2005
    #5
  6. 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
    Uno Qualunque (fu: Fil), Feb 1, 2005
    #6
  7. Mark De Biasi

    HubSwitch Guest

    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" <> wrote in message
    news:...
    > 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?
    HubSwitch, Feb 1, 2005
    #7
  8. Mark De Biasi

    m Guest

    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
    m, Feb 2, 2005
    #8
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