Alsa errors: is this pjsip bug?

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On Fri, Aug 1, 2014 at 8:41 AM, Warpme <warpme at o2.pl> wrote:
> On 31/07/14 20:05, Ernesto Celis wrote:
>>
>> It looks like the audio backend is unable to set the appropriate bit
>> rate,  try building pjsip disabling PortAudio and enabling Alsa
>> backend instead.
>
> Thx Ernesto!
>
> I'm looking on ./configure and don't see how to disable portaudio and enable
> alsa
> May You hint me how to do this?

A quick search in duckduckgo (or google if you like it more) "pjsip
disable portaudio" returns many links where you can see how to disable
PortAudio back end and to enable Alsa instead. Please, do your own
research before asking in the mailing list, your chances of getting
help will increase if you do.

Read pjsip documentation about the file config_site.h which is the
place to configure some settings besides the configure script.This
link http://trac.pjsip.org/repos/wiki should become your holy book
regarding pjsip.

>
> I tried current pjproject-svn (r4883) build with external portaudio
> (svn1919).

I don't use the svn code, I prefer to stick to the releases, currently at 2.2.1

> There seems to be small progress, as now log looks following:
>
> I understand above set of "Expression....failed...." is because psjip tries
> to set sound-dev to PCM at 16000 - and this isn't supported on my hardware.

Yes there is a problem to set the appropriate bit rate for your
hardware, but since I don't know your hardware I can't tell for sure
where the problem is.

> 15:32:28.198    pjsua_aud.c  ...Conf connect: 1 --> 0
> 15:32:28.198   conference.c  ....Port 1 (sip:508047160 at 192.168.1.254)
> transmitting to port 0 (HDA NVidia: ALC889A Analog (hw:0,0) (44KHz))
> 15:32:28.198    pjsua_aud.c  ...Conf connect: 0 --> 1
> 15:32:28.198   conference.c  ....Port 0 (HDA NVidia: ALC889A Analog (hw:0,0)
> (44KHz)) transmitting to port 1 (sip:508047160 at 192.168.1.254)

Per the log the conference bridge successfully connects the incoming
and outgoing audio, but at 44KHz, which your hardware is unable (it
seems) to be able to handle. But you should check first if it works
with the Alsa back end. Have you ran the pjsystest application which
is built along with the library? You should find it under
pjsip-apps/bin IIRC.

There is only one sound device on your hardware? Somewhere in your log
output I see something about setting sound device. Try without
manually setting the devices (in case you were doing it, if not just
ignore this), pjsip will use the first sound device it finds.

> All above looks all OK for me.
> Unfortunately local->remote audio still isn't working.

There is a section in the wiki devoted to troubleshoot audio related
issues, you should check it.

> Instead of, I have local MIC looped to local speakers.

What do you mean? Are you sure you aren't connecting the mic audio to
the audio playback only? Check this
http://trac.pjsip.org/repos/wiki/Python_SIP/Media to get a better
understanding of how pjsip handles audio.

> Where issue might be?

How can I know? My crystal ball is broken, I'm sorry :)

> BTW: Infact I hear local MIC all the time in local speakers - no difference
> is SIP app launched or not.
> Is this normal?

Yes, that is how sound in computers work. Be sure you are not running
pulseaudio and that no other application is trying to use the audio
device.

In your first email you said you were running some kernel version, it
is more useful if you say which distribution are you running instead,
the kernel version is very much useless here.

-- 
Saludos

Ernesto Celis de la Fuente
http://expressit.celisdelafuente.net



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