No Voice Flow

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



Hi Smita,

Yes the forums are very true in this regard infact I had also faced the same
problem on the emulator. The emulator has the half duplex issue. For these
reasons i had told you can better use some real device for testing. I am
sure that this issue will not come on the device.

Also take a guess your logs and half duplex issue are also interrelated. see
symb_mda_dev.c is nothing but the implementation if Audio device API and the
log error specifies that it has some problem in recording the input/output
media stream and also sue to which there is buffer overrun. hence the voice
is half duplex.

also #define PJMEDIA_AUDIO_DEV_HAS_SYMB_MDA   PJ_SYMBIAN
This setting controls whether Symbian audio (using built-in multimedia
framework) support should be included. This enables the built in symbian
audio support on emulator which may have issues in implementaion in emulator
thyself.

So from the above discusion you can conclude tha either ignore this issue
and can continue for further development and if it is critical to be
resolved first as per development strategy then you may use the device some
how and test. and the last but not the least wait for some wizard of symbian
to put some comments on  it.


Regards;
Varun Pratap singh,
SIP Infrastructure Expert,
Mob: 91-9873868905,
Skype: varunps2003


On Thu, Jul 16, 2009 at 9:09 AM, Smiti Gupta <Smiti_Gupta at infosys.com>wrote:

>  Hi Varun,
>
>
>
> I am using S60 3rd FP2 SDK, PJSIP version 1.1 and currently trying on
> emulator. The information I got from various forums is that half-duplex
> works on emulator so atleast one party should be able to hear. I have also
> tried a call between Xlite and emulator. In that case, I am able to hear the
> person speaking from Xlite but xlite person is not able to hear anything.
>
>
>
> Also in the logs I m getting a message: symb_mda_dev.c Error in
> MaiscRecordComplete(): error code -14 and lots of ?Master/sound Underflow,
> buf_cnt=0, will generate 1 frame? messages.
>
>
>
> Your help will be really appreciated.
>
>
>
> Thanks & Regards
>
> Smiti
>  ------------------------------
>
> *From:* pjsip-bounces at lists.pjsip.org [mailto:
> pjsip-bounces at lists.pjsip.org] *On Behalf Of *varun pratapsingh
> *Sent:* Wednesday, July 15, 2009 3:02 PM
> *To:* pjsip list
> *Subject:* Re: [pjsip] No Voice Flow
>
>
>
> Hi Smiti,
>
> Can I get some more detail. you are using the device or emulator. If you
> are using the emulator then please test it on the real device because the
> emulator has been reported some incapabilties regarding some capabilities.
> Also specify the SDK you are using.
>
> I can really help you in this because I have already developed a Softphone
> for Symbian OS successfully.
>
>
> Regards:
> Varun Pratap Singh,
> SIP Infrastructure Expert,
> Mob: 91-9873868905,
> Skype: varunps2003
>
>  On Fri, Jul 10, 2009 at 9:30 AM, Smiti Gupta <Smiti_Gupta at infosys.com>
> wrote:
>
> Hi All,
>
>
>
> I am developing a softphone for Symbian. I am able to establish a call
> between the two softphone clients, but there is no voice flow.
>
> Both the parties are not able to hear anything.
>
>
>
> Does anyone have any idea how to fix this or where the problem could be?
>
>
>
> Thanks & Regards
>
> Smiti
>
> **************** CAUTION - Disclaimer *****************
>
> This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely
>
> for the use of the addressee(s). If you are not the intended recipient, please
>
> notify the sender by e-mail and delete the original message. Further, you are not
>
> to copy, disclose, or distribute this e-mail or its contents to any other person and
>
> any such actions are unlawful. This e-mail may contain viruses. Infosys has taken
>
> every reasonable precaution to minimize this risk, but is not liable for any damage
>
> you may sustain as a result of any virus in this e-mail. You should carry out your
>
> own virus checks before opening the e-mail or attachment. Infosys reserves the
>
> right to monitor and review the content of all messages sent to or from this e-mail
>
> address. Messages sent to or from this e-mail address may be stored on the
>
> Infosys e-mail system.
>
> ***INFOSYS******** End of Disclaimer ********INFOSYS***
>
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
> pjsip at lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
>
>
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
> pjsip at lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20090716/c4e017f6/attachment.html>


[Index of Archives]     [Asterisk Users]     [Asterisk App Development]     [Linux ARM Kernel]     [Linux ARM]     [Linux Omap]     [Fedora ARM]     [IETF Annouce]     [Security]     [Bugtraq]     [Linux]     [Linux OMAP]     [Linux MIPS]     [Linux API]
  Powered by Linux