CODEC

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

 



On Tue, Nov 3, 2009 at 7:17 PM, Kresten Tolstrup <kt at danphone.com> wrote:
> Hi,
>
> Isn't there anyone that kan help me with this problem? The short version of
> the problem is that I need a passthrought CODEC, but when I use a
> passthrought CODEC, the other end don't get any CODEC information, in the
> SIP call.
>

Snippet of message trace would help here.

But actually, we do have a sample to use hardware codec on Windows!
The native WMME audio backend supports PCMA/PCMU codec, and here's
what I have in my config_site.h to enable it:


#define PJ_CONFIG_WIN32_WMME_DIRECT	1
#include <pj/config_site_sample.h>

#define PJMEDIA_HAS_L16_CODEC		0
#define PJMEDIA_HAS_G711_CODEC		0
#define PJMEDIA_HAS_GSM_CODEC		0
#define PJMEDIA_HAS_ILBC_CODEC		0
#define PJMEDIA_HAS_G722_CODEC		0
#define PJMEDIA_HAS_SPEEX_CODEC		0
#define PJMEDIA_HAS_G7221_CODEC		0

I just tested with latest pjsua and it seems to work.

Cheers
 Benny





> Best Regards
> Kresten
>
> -----Oprindelig meddelelse-----
> Fra: pjsip-bounces at lists.pjsip.org
> [mailto:pjsip-bounces at lists.pjsip.org]P? vegne af Kresten Tolstrup
> Sendt: 30. oktober 2009 14:15
> Til: pjsip list
> Emne: Re: [pjsip] CODEC
>
>
> Hi Naning
>
> Thanks for your answer. I mean PCMA.
>
> I have read the link you gave me, but I have still problems.
>
> I have added #define PJMEDIA_CONF_USE_SWITCH_BOARD 1 in config_site.h and
> the passthrough codec for PCMA is also set to 1, but when I make a call to
> the PJSIP stack, I get the following error in the SIP SPD: "No suitable
> codec for remote offer (PJMEDIA_SPDNEG_NOANSCODEC)". It looks like the SIP
> stack don't have any passthrough codec, although #define
> PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMA is set to 1. All codec defines are set to
> zero.
>
> I have looked in the function pjsua_media_subsys_init, in file
> pjsua_media.c, under the define PJMEDIA_HAS_PASSTHROUGH_CODECS. When I run
> the code it finds no sound device, to me it looks like that could be the
> problem, because ext_fmt_cnt remains on zero, how to add a dummy sound
> device, I tried with null_sound_device, but it doesn't work.
>
> The samples shall be transmitted / received over a SSC bus, where the codec
> is connected, so I have made my own PJSIP port, where the to functions put-
> and getframe transmit and receive the data from and to the SSC bus, so I
> don't need any sounddevice.
>
> Does anyone know how to solve this problem?
>
> Best Regards
> Kresten
>
> -----Oprindelig meddelelse-----
> Fra: pjsip-bounces at lists.pjsip.org
> [mailto:pjsip-bounces at lists.pjsip.org]P? vegne af Nanang Izzuddin
> Sendt: 29. oktober 2009 17:52
> Til: pjsip list
> Emne: Re: [pjsip] CODEC
>
>
> Hi Kresten,
>
> Did you mean PCMU/u-law hardware codec?
>
> Please also note that enabling passthrough codecs will require audio
> switchboard as described in APS/VAS-direct wiki [1].
>
> Log snippet could be very useful.
>
> ---
> [1] http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct
>
> BR,
> nanang
>
>
> On Thu, Oct 29, 2009 at 8:07 PM, Kresten Tolstrup <kt at danphone.com> wrote:
>> Hi,
>>
>> I have a a-low hardware codec together with a WinCE uP. Therefore I want
> to
>> use PJSIP with a passthrough CODEC. in the file config_site.h I have set
> the
>> following:
>>
>> #define PJMEDIA_HAS_PASSTHROUGH_CODECS ?1
>> #define PJMEDIA_HAS_L16_CODEC ? ? ? ? ? 0
>> #define PJMEDIA_HAS_ILBC_CODEC ? ? ? ? ?0
>> #define PJMEDIA_HAS_GSM_CODEC ? ? ? ? ? 0
>> #define PJMEDIA_HAS_SPEEX_CODEC ? ? ? ? 0
>> #define PJMEDIA_HAS_G722_CODEC ? ? ? ? ?0
>> #define PJMEDIA_HAS_PASSTHROUGH_CODEC_G729 ?0
>> #define PJMEDIA_HAS_PASSTHROUGH_CODEC_ILBC ? ? ?0
>> #define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMA ? ? ?0
>>
>> When I make a call to SJphone, and PJMEDIA_HAS_PASSTHROUGH_CODECS is set
> to
>> 1, SJphone says that there is CODEC error, but when
>> PJMEDIA_HAS_PASSTHROUGH_CODECS is set to 0 the sip call go through.
>> How can I fix this?
>>
>> Best regards
>> Kresten
>>
>>
>> _______________________________________________
>> 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
>
>
> _______________________________________________
> 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
>



[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