Re: Fw: Sound issues: strange samplerates?

Bill Gardner <billg@xxxxxxxxxxxx> · Wed, 6 Apr 2016 16:52:00 -0400



    Hi Oliver,

    
    Please generate a pjsip logfile (level 4 should suffice) and send,
    there may be clues in there.

    
    Bill

    
    On 4/6/2016 4:26 PM, Oli Kah wrote:

    
          Hi Bill, hi everyone,
           
          thanks for answering but I can confirm that the
            config_site.h settings are NOT the problem. I have now
            recompiled the lib with nothing in config_site.h.
          The result is exactly the same.
           
          To illustrate what the called person hears I have
            attached an mp4 which also shows the progress of the Python
            app in form of PyCharms debug log.
           
          There are important issues that can be seen + heard. Note
            that I am saying "1-2-3" from the beginning(!) of the video
            until the very end without ever stopping. Only the receiving
            side can be heard voice-wise (or not as you will see/hear).
           
          1) When picking up the phone the sound CANNOT be heard on
            the receiving side although I am continously saying "1-2-3".
            It takes roughly 10(!!) seconds before the other side can
            hear me at all!!
          2) It also takes 10 seconds after pickup before the call
            confirmation phase is reached. This is extremly slow and
            totally unexpected.
          3) When the other side finally can hear me I sound like a
            monster... the sample rate seems to be off - I can't
            otherwise explain the strange sounding voice!
           
          The Python code of this mini app is attached to this
            email once more.
           
          I don't think I am doing anything exotic. It does not
            work as expected though. If I do the same with an app like
            Phoner (see http://www.phoner.de/download_en.htm) none of
            these problems occur within the same environment and using
            the same accounts and phone numbers. So it has to do with
            pjsip lib somehow.
           
          Any ideas?
           
          Thank you.
           
          Cheers,
          
            Oliver
             
            
              Gesendet: Montag,
                04. April 2016 um 20:49 Uhr

                Von: "Bill Gardner" <billg@xxxxxxxxxxxx>

                An: pjsip@xxxxxxxxxxxxxxx

                Betreff: Re:  Fw: Sound issues: strange
                samplerates?
              
                Hi
                  Oliver,

                  
                  I think you should try a completely default
                  configuration, i.e. use an empty config_site.h file.
                  Your config_site.h params may be causing problems.

                  
                  Regards,

                  
                  Bill

                   
                  On 4/4/2016 2:37 PM, Oli
                    Kah wrote:
                  
                    
                        Hmmm, no one?
                         
                        Is there some sort of forum somewhere where
                          to post things like these??
                         
                        Thank you :)
                         
                        Cheers,
                        Oli
                         
                          
                            Gesendet: Freitag,
                              01. April 2016 um 21:38 Uhr

                              Von: "Oli Kah" <mj_fn@xxxxxx>

                              An: pjsip@xxxxxxxxxxxxxxx

                              Betreff: Sound issues: strange
                              samplerates?
                            
                              
                                Hi there,
                                 
                                 
                                I am new to this list and want to
                                  say "Hello" to everyone listening :)
                                 
                                My issue using pjsib is rather
                                  strange. When calling someone (I
                                  tested it using two of my own
                                  telephone numbers) I get normal audio
                                  first during early call stages and
                                  then when the call is finally
                                  confirmed the audio rate suddenly is
                                  half or so. The voice then sounds
                                  monsterish and the remote site is no
                                  longer audible via speakers. This
                                  happens every time - reproducible!
                                 
                                What I did:
                                 
                                I have successfully compiled pjsua
                                  (release build) for Python using
                                  Visual Studio 2015 Community linking
                                  pjsua to Python 2.6 lib, 32bit running
                                  everything on Windows 8.1 Pro 64bit.
                                  So far so good.
                                These values were used for my
                                  config_site.h:
                                 
                                #define
                                  PJMEDIA_CONF_USE_SWITCH_BOARD 1

                                  #define PJMEDIA_SOUND_BUFFER_COUNT 8

                                  #define
                                  PJMEDIA_SND_DEFAULT_REC_LATENCY 50

                                  #define
                                  PJMEDIA_SND_DEFAULT_PLAY_LATENCY 50
                                 
                                The attached python code shall
                                  define a simple SIPPhone that will be
                                  used in my larger scale application.
                                  It's unfinished but I ran into these
                                  unresolvable audio problems and hope
                                  that you might help me. In its current
                                  form the SIPPhone is totally useless!
                                 
                                During testing I have been using
                                  the onboard audio card of my mainboard
                                  with some Bose speakers and a Samson
                                  UB-1 USB microphone which is connected
                                  to the Windows-PC using an USB 2.0
                                  port. For the "remote" side I use a
                                  dedicated VOIP telephone from
                                  Grandstream.
                                 
                                The test code in my attached file
                                  is then run using Pycharm + Python 2.6
                                  runtimes. For testing you must replace
                                  the numbers + credientials by valid
                                  values on your side. Also note that
                                  you might have to specify another
                                  domain (mine here is "fritz.box").
                                 
                                The called Grandstream rings.
                                  Taking up the phone then starts the
                                  call confirmation which is going
                                  unexpectedly slow. It takes quite some
                                  time (between roughly 3-10 seconds)
                                  before the "confirmed" status is
                                  reached although the audio starts
                                  working before that. The scenario is
                                  running on my LAN where the SIP server
                                  (a FritzBox) is also running here. So
                                  quite strange why this takes so long?
                                Right after phone pickup the audio
                                  can't be heard at all (me constantly
                                  talking after phone pickup!) - in both
                                  directions! Then after a short time
                                  (1-4 sec) the voice can be heard
                                  normally and understandably as
                                  expected in both directions. But this
                                  only works until the "confirmed"
                                  status is reached for the call. When
                                  it is reached there is no longer any
                                  sound coming from the PC speaker and
                                  the voice heard on the Grandstream is
                                  monsterish (half sample rate?!). The
                                  bidirectional audio is lost and the
                                  remaining audio is really bad.
                                 
                                The observed behavior happens every
                                  time. Just the timing is different and
                                  the time from starting the program to
                                  hearing the monsterish voice is
                                  between 3 and 10 seconds. It is also
                                  rather strange that this time span can
                                  be so huge!
                                 
                                 
                                What happens here?! In its current
                                  form the SIPPhone is pretty unusable.
                                  I hope you have ideas how to fix that
                                  :) I have experimented with many of
                                  the MediaConfig parameters with no
                                  success.
                                 
                                 
                                Thank you!
                                 
                                Cheers,
                                Oliver
                                 
                              
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