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