[pjsip] Fw: Sound issues: strange samplerates?

Bill Gardner billg at wavearts.com
Mon Apr 4 14:49:44 EDT 2016

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.



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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