[pjsip] CODEC
Kresten Tolstrup
kt at danphone.com
Tue Nov 3 06:17:26 CST 2009
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.
Best Regards
Kresten
-----Oprindelig meddelelse-----
Fra: pjsip-bounces at lists.pjsip.org
[mailto:pjsip-bounces at lists.pjsip.org]På 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]På 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
More information about the pjsip
mailing list