PJSUA jitter values
Hello,
I was monitoring the jitter values of pjsua using Ethreal and the mean
jitter value was 15ms. I have monitored other sip soft phones e.g. Xlite and
SJPhone and their jitter values are much lower then this. Using >>qd on the
console shows different jitter values than those shown by the Ethereal.
These are usually high compared to Ethereal.
I am using the G .711 PCMA codec.I have also tried to change the jitter
buffer settings, i.e. reduced the initial, minimum and maximum jitter buffer
size values but these changes do not affect the resulting mean jitter at
all! Is there anyway to reduce the jitter delay on pjsua phone? Awaiting you
response.
--
Regards,
Salman
On 10/25/07, Salman, Muhammad ms.salman@gmail.com wrote:
all! Is there anyway to reduce the jitter delay on pjsua phone? Awaiting you
response.
Have you tried the steps here: http://www.pjsip.org/trac/wiki/FAQ#audio-latency
HTH
--
Perry Ismangil
Yes, I did follow these steps but it would seem that lowering the audio
On 10/25/07, Perry Ismangil perry@pjsip.org wrote:
On 10/25/07, Salman, Muhammad ms.salman@gmail.com wrote:
all! Is there anyway to reduce the jitter delay on pjsua phone? Awaiting
you
response.
Have you tried the steps here:
http://www.pjsip.org/trac/wiki/FAQ#audio-latency
HTH
--
Perry Ismangil
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
--
Regards,
Salman
Yes, I did follow these steps but it would seem that lowering the audio
frame length and setting the to 6 instead of 16 increased the jitter
further. I have no idea on how to change the values of the jitter buffer so
I kept them at defaults of 60, 240, 360 and 150. Would someone please
explain how to set these values / recommend a set of values that might work?
On 10/25/07, Perry Ismangil perry@pjsip.org wrote:
On 10/25/07, Salman, Muhammad ms.salman@gmail.com wrote:
all! Is there anyway to reduce the jitter delay on pjsua phone? Awaiting
you
response.
Have you tried the steps here:
http://www.pjsip.org/trac/wiki/FAQ#audio-latency
HTH
--
Perry Ismangil
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
--
Regards,
Salman
Salman, Muhammad wrote:
Yes, I did follow these steps but it would seem that lowering the audio
frame length and setting the to 6 instead of 16 increased the jitter
further. I have no idea on how to change the values of the jitter buffer
so I kept them at defaults of 60, 240, 360 and 150. Would someone
please explain how to set these values / recommend a set of values that
might work?
All the jitter settings are in pjsua_media_config.
Reading your mails, it seems that you want is to reduce the
transmission jitter, in this case the jitter buffer settings won't
have any effects since they are for accommodating reception jitter.
FYI outgoing transmission jitter is explained in
http://www.pjsip.org/trac/wiki/FAQ#tx-timing.
cheers,
-benny
On 10/25/07, Perry Ismangil <perry@pjsip.org mailto:perry@pjsip.org>
wrote:
On 10/25/07, Salman, Muhammad <ms.salman@gmail.com
<mailto:ms.salman@gmail.com>> wrote:
all! Is there anyway to reduce the jitter delay on pjsua phone?
Awaiting you
response.
Have you tried the steps here:
http://www.pjsip.org/trac/wiki/FAQ#audio-latency
HTH
--
Perry Ismangil
Hi,
I made the changes suggested in
http://www.pjsip.org/trac/wiki/FAQ#pjsua-lib-perf.
I'm using the following parameters:
for the conference bridge:
Conference bridge slots: 10
Sample rate: 8000 Hz
Channels: 1
samples per frame: (20*Sample rate)/1000
Bits per sample: 16
Bridge Option: PJMEDIA_CONF_NO_DEVICE |
PJMEDIA_CONF_USE_LINEAR
for the application:
pjsua_config::max_calls = 500
my config_site.h has the options inside
config_site_sample.h under PJ_CONFIG_MAXIMUM_SPEED
#define
I made a application that receives calls and plays a
wav file. With a few calls it goes ok. If I try to put
a lot of calls I get some problems. After I get 373
calls the application quits and the function
pjmedia_master_port_create returns the code 120012. If
I print it with pj_strerror() it says: "Cannot
allocate memoryAborted".
All help is very welcome,
Thiago
Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento!
I'm doing some more tests and found some strange
things (most probably my code...). I'm using every
media port with the same parameters (8 kHz, 16 bit,
160 samples per frame). When I connect (dial in) to
the application I hear the welcome wav with no
problems (played when call media gets active). Now if
I try to add the audio stream from one call to the
conference bridge of another call (to put them in
conference for example), the audio in the first call
(the one that I grabbed the audio stream) get really
horrible (slows down, cuts a lot of frames...), even
before I connect both calls in the conference bridge.
If I remove the port from the destination conference
bridge, the original call gets perfect again.
Thats one thing. Another one is if I hangup the call
that I got the audio stream and added to another
conference bridge, I get
../src/pj/os_core_unix.c:1048: pj_mutex_lock:
Assertion `mutex' failed and the application closes.
If I put any call on hold (destroy media), no mater if
it was added to another conference bridge or not, I'll
get a seg fault and the following stack:
8 put_frame_imp() 0x080a1f69
7 put_frame() 0x080a24a0
6 get_frame() 0x080981f3
5 clock_callback() 0x0809a52f
4 clock_thread() 0x080a7f01
3 thread_main() 0x080ebcf7
2 start_thread() 0xb7ebd240
1 clone() 0xb7b4f4ae
My application is object oriented, and I have one
object per call, and this class has one conference
bridge for each call. I'm using a null sound port. I
already tried to change the conference bridge
resampling to see if it changes the quality, but no
difference (as expected, since I'm using all sampling
rates equal). All my call are using pcmu codec. My
conference bridges have 10 max slots in each.
It seems like there is a problem to add the call audio
stream to more than one conference bridge. I may be
missing something here... not sure what.
My system is a Debian Linux 4.0 (Etch) and I'm using
pjsip from 25-10-2007 (I think its r 1522).
Thanks a lot for any help, I'm pretty lost here...
Thiago Paiva Flores
Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento!
Thank you for your insight. It was really helpful.
You suggested if I wanted to reduce the transmission jitter. Yes, that is
exactly what I wanted. But installing a master port between the conference
bridge and sound device doesn't seem to have any effect in this case. The
transmission jitter still remains the same. I am attaching snap shots of the
forward (pjsua) and reverse (some other sip phone) RTP stream analysis using
Wire Shark.
It seems that during transmission the delta for the forward stream
fluctuates in contrast to the reverse stream whose delta and jitter remain
pretty much the same.
Thank you for your time.
On 10/26/07, Benny Prijono bennylp@pjsip.org wrote:
Salman, Muhammad wrote:
Yes, I did follow these steps but it would seem that lowering the audio
frame length and setting the to 6 instead of 16 increased the jitter
further. I have no idea on how to change the values of the jitter buffer
so I kept them at defaults of 60, 240, 360 and 150. Would someone
please explain how to set these values / recommend a set of values that
might work?
All the jitter settings are in pjsua_media_config.
Reading your mails, it seems that you want is to reduce the
transmission jitter, in this case the jitter buffer settings won't
have any effects since they are for accommodating reception jitter.
FYI outgoing transmission jitter is explained in
http://www.pjsip.org/trac/wiki/FAQ#tx-timing.
cheers,
-benny
On 10/25/07, Perry Ismangil <perry@pjsip.org mailto:perry@pjsip.org>
wrote:
On 10/25/07, Salman, Muhammad <ms.salman@gmail.com
<mailto:ms.salman@gmail.com>> wrote:
all! Is there anyway to reduce the jitter delay on pjsua phone?
Awaiting you
response.
Have you tried the steps here:
http://www.pjsip.org/trac/wiki/FAQ#audio-latency
HTH
--
Perry Ismangil
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
--
Regards,
Salman
Salman, Muhammad wrote:
Thank you for your insight. It was really helpful.
You suggested if I wanted to reduce the transmission jitter. Yes, that
is exactly what I wanted. But installing a master port between the
conference bridge and sound device doesn't seem to have any effect in
this case. The transmission jitter still remains the same. I am
attaching snap shots of the forward (pjsua) and reverse (some other sip
phone) RTP stream analysis using Wire Shark.
I suspect PortAudio's thread has interfered with the clock's thread,
since PortAudio's thread priority is set to real-time. Could you try
increasing the pjmedia clock thread's priority to real-time too, in
clock_thread.c?
-benny
It seems that during transmission the delta for the forward stream
fluctuates in contrast to the reverse stream whose delta and jitter
remain pretty much the same.
Thank you for your time.
Hi everyone,
I'm using pjsip 0.5.10.4 in an equipment that as three Ethernet
devices each one with a IP address. I want to create 2 accounts for
Eth1, 2 accounts for Eth2 and none for Eth0, using the same PJSUA. Is
it possible?
Thanks everyone,
Jorge Amador