Additional results from testing.
Tried different settings in config_site.h:
#define PJMEDIA_HAS_PASSTHROUGH_CODEC_AMR 0
#define PJMEDIA_HAS_PASSTHROUGH_CODEC_G729 0
#define PJMEDIA_HAS_PASSTHROUGH_CODEC_ILBC 1
From these three only AMR is working on S60 3rd and 5th without problems.
ILBC crashes the app every time (on 5th edition X6). While G729 doesn't
show any problems, there's no sound transmitted (checked with wireshark).
I was also testing with these params:
# define PJMEDIA_AUDIO_DEV_SYMB_VAS_VERSION 1
# define PJMEDIA_AUDIO_DEV_SYMB_APS_DETECTS_CODEC 2
But with the same results.
If PJMEDIA_AUDIO_DEV_SYMB_VAS_VERSION is defined as 2 (S60 5th?). Pjsip
does not compile (again - I'm using v1.6): "EG711_10MS undeclared (first
use this function) symb_vas_dev.cpp"
What am I missing?
Tine Ur?i? wrote:
> In the meantime I did some research. Wireshark told me that I'm using
> speex. No Idea how it got there though (disabled on the priority list).
> Now I disabled codecs in config_site.h:
>
> /* Select codecs to disable */
> #define PJMEDIA_HAS_L16_CODEC 0
> #define PJMEDIA_HAS_ILBC_CODEC 0
> #define PJMEDIA_HAS_G722_CODEC 0
> #define PJMEDIA_HAS_G711_CODEC 0
> #define PJMEDIA_HAS_GSM_CODEC 0
> #define PJMEDIA_HAS_SPEEX_CODEC 0
> #define PJMEDIA_HAS_INTEL_IPP 0
>
>
> After the change I can't make a call with nokia X6. On an older N78 it
> seems OK.
> Error (nothing specific, app crashes):
> Process 1575, thread 1576 stopped at 0x802c520f: Thread 0x628 has
> panicked. Category: KERN-EXEC; Reason: 0
>
> It happens somewhere inside "pjsua_conf_connect(CallDetails.conf_slot,
> 0) ;". Any ideas?
>
>
>
> Tine Ur?i? wrote:
>> Hi.
>>
>> Thanks for the response.
>>
>> I'm using VAS-Direct (in config_site.h: "#define
>> PJ_CONFIG_NOKIA_VAS_DIRECT 1") . The codec should be ilbc - I
>> disabled the other codecs and put ilbc on the top priority
>> (pjsua_codec_set_priority), but I'm not completely sure it was
>> correctly negotiated.
>> How can I check for the codec being used during call?
>>
>> What exactly did you mean by "Try pjmedia_conf_get_port_info() for
>> the destination port."? I did try to use pjsua_conf_get_port_info to
>> get some more info on the stream:
>>
>> pjsua_conf_port_info PortInfo;
>>
>> status = pjsua_conf_get_port_info( pjsua_call_get_conf_port(CallID), &PortInfo );
>> if (status != PJ_SUCCESS)
>> {
>> qDebug() << "Error while reading port info:" << status;
>> }
>>
>> Debug output:
>> PortInfo name: sip:xxxxxxx at xx.xx.xx.xx:5060;user=phone
>> PortInfo clock rate: 16000
>> PortInfo channel count: 1
>> PortInfo samples per frame: 320
>> PortInfo bits per sample: 16
>> PortInfo listener cnt: 0
>>
>>
>> Best regards,
>> Tine
>>
>>
>> Nanang Izzuddin wrote:
>>> Hi,
>>>
>>> Try pjmedia_conf_get_port_info() for the destination port.
>>>
>>> Are you using VAS-Direct? And which codec was being used by the call stream?
>>>
>>> BR,
>>> nanang
>>>
>>>
>>> On Wed, Oct 13, 2010 at 2:11 PM, Tine Ur?i? <tine at cde.si> wrote:
>>>
>>>> Hi.
>>>>
>>>> I'm having a bit of trouble with playing a wave audio file to caller while
>>>> "on hold".
>>>>
>>>> The error returned by pjsua_conf_connect is "220161" (Incompatible clock
>>>> rate - PJMEDIA_ENCCLOCKRATE).
>>>> The manual states: "WAV player port supports for reading WAV file with
>>>> uncompressed 16 bit PCM format or compressed G.711 A-law/U-law format.".
>>>> I've tried a lot of different wave formats (including 16bit 8khz standard)
>>>> and I can't seem to find the right wave settings.
>>>> If I connect player to the local speaker it works just fine (example:
>>>> pjsua_conf_connect(pjsua_player_get_conf_port(player_id), 0); ).
>>>>
>>>> I'm using pjsip v1.6 on a symbian S60 device (VAS).
>>>>
>>>> How can I find which clock rate to use?
>>>>
>>>> Here's my sample code:
>>>>
>>>> pj_status_t status ;
>>>> pjsua_call_info CallDetails ;
>>>> pjsua_player_id player_id;
>>>>
>>>> if (pjsua_call_get_info(CallID, &CallDetails) == PJ_SUCCESS)
>>>> {
>>>> //disconnect caller
>>>> pjsua_conf_disconnect(CallDetails.conf_slot, 0) ;
>>>> pjsua_conf_disconnect(0, CallDetails.conf_slot) ;
>>>>
>>>> //create player
>>>> pj_str_t awav = pj_str ("C:\\Data\\OnHold.wav");
>>>> status = pjsua_player_create(&awav, 0, &player_id);
>>>> if (status != PJ_SUCCESS)
>>>> {
>>>> status =
>>>> pjsua_conf_connect(pjsua_player_get_conf_port(player_id),
>>>> CallDetails.conf_slot);
>>>> ...
>>>> }
>>>>
>>>> }
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>
>>
>>
>> --
>> Tine Ur?i?
>> Razvojni in?enir
>>
>> CDE nove tehnologije d.o.o.
>> Tehnolo?ki park 24
>> 1000 Ljubljana, Slovenija
>>
>> GSM: +386 0 30 600 619
>> E-mail: tine at cde.si
>> www.cde.si
>
>
> --
> Tine Ur?i?
> Razvojni in?enir
>
> CDE nove tehnologije d.o.o.
> Tehnolo?ki park 24
> 1000 Ljubljana, Slovenija
>
> GSM: +386 0 30 600 619
> E-mail: tine at cde.si
> www.cde.si
--
Tine Ur?i?
Razvojni in?enir
CDE nove tehnologije d.o.o.
Tehnolo?ki park 24
1000 Ljubljana, Slovenija
GSM: +386 0 30 600 619
E-mail: tine at cde.si
www.cde.si
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