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
