[pjsip] Fw: Sound issues: strange samplerates?

Bill Gardner billg at wavearts.com
Wed Apr 6 16:52:00 EDT 2016

Hi Oliver,

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


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 at wavearts.com>
> *An:* pjsip at lists.pjsip.org
> *Betreff:* Re: [pjsip] 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 at web.de>
>     *An:* pjsip at lists.pjsip.org
>     *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:
>     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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