Voice quality with software SIP phone`

Discussion in 'General' started by ericng, Apr 15, 2009.

  1. ericng

    Joined:
    May 14, 2008
    Messages:
    78
    Likes Received:
    0
    When used with eyeBeam from CounterPath, we found there always some background sound which we are unable to remove. Is this something to do with Codec code that been applied or there is any needs to upgrade the codec?

    BTW, is there anyone has a good voice quality softphone to recommend?

    Thanks

    Eric
     
  2. Chilling_Silence

    Joined:
    Sep 23, 2008
    Messages:
    488
    Likes Received:
    0
    We find varying results with different softphones.

    Common ones to try:
    X-Lite
    ZoIPer
    QuteCom (Previously OpenWengo)

    Mix and match and one will suit your needs :)
     
  3. mattrh

    Joined:
    Jul 15, 2008
    Messages:
    175
    Likes Received:
    0
    what i have found most of the time it is just feedback from the mic. but just as a precautions check your ping times, packet dropping, bandwidth allocation and type of connection , and make sure you have QoS on your network if you have a lot of data on a single switch.
     
  4. ramoncio

    Joined:
    May 12, 2010
    Messages:
    1,663
    Likes Received:
    0
    Use a low compression codec as g711a/u and QOS is a must if you want to avoid sound quality problems.
     
  5. danardf

    Joined:
    Dec 3, 2007
    Messages:
    8,069
    Likes Received:
    12
    What kind of sound?
    Echo, breath, communication discontinued ?

    Maybe you have your mic gain to high.
    Do you use your sofphone with headset or handset or free hands?
     
  6. ericng

    Joined:
    May 14, 2008
    Messages:
    78
    Likes Received:
    0
    I thought we should use as high compression codec to achieve the good sound quality? BTW, where can we get these codec as some of them are not free.

    Eric
     
  7. Patrick_elx

    Joined:
    Dec 14, 2008
    Messages:
    1,120
    Likes Received:
    0
    There are two concerns:

    - limiting the bandwidth used on your internet link (if you are using VoiP trunk). For instance if you are using DSL with an upload BW of 512k, you probably won't be able to have more than four or five g711 com at the same time and if nobody else is using the internet at the same time. You will have then a loss of quality due to loss of packet, time delay etc... To avoid these problem we are using codec that are more compressed (ilbc, g729, gsm etc...)


    - To provide the best audio quality, you want a codec that will not lose too much data when compressing. g711 is far better than g729 or gsm. The tradeoff is that the better they are, the more bandwidth they are using. The more compressed, the more artifact noise you can also introduce due to sampling errors.

    Now depending on what are your limiting factors on your system you will have to make some choice:

    high bandwidth, not a lot of simultaneous calls: go for g711
    low bandwidth or a lot of simultaneous calls: g729 or gsm will probably be the only things available.

    In between: that's where you can start playing around..
     
  8. ramoncio

    Joined:
    May 12, 2010
    Messages:
    1,663
    Likes Received:
    0
    Nice explanation Patrick!
     

Share This Page