Hi,
i have successfully compiled the pjsip(V1.6) with APS support and have been
testing the symbian_ua_gui on my nokia E66 with my asterisk server . But it
seems that the calls are always made in g711 .
codec. I have tried to force the g729 in my config_site_sample.h header file
in the following manner:
#ifdef PJ_CONFIG_NOKIA_APS_DIRECT
/* MUST use switchboard rather than the conference bridge */
#define PJMEDIA_CONF_USE_SWITCH_BOARD 1
/* Enable APS sound device backend and disable MDA & VAS */
#define PJMEDIA_AUDIO_DEV_HAS_SYMB_MDA 0
#define PJMEDIA_AUDIO_DEV_HAS_SYMB_APS 1
#define PJMEDIA_AUDIO_DEV_HAS_SYMB_VAS 0
/* Enable passthrough codec framework */
#define PJMEDIA_HAS_PASSTHROUGH_CODECS 1
/* And selectively enable which codecs are supported by the handset */
#define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMU 0
#define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMA 0
#define PJMEDIA_HAS_PASSTHROUGH_CODEC_AMR 0
#define PJMEDIA_HAS_PASSTHROUGH_CODEC_G729 1
#define PJMEDIA_HAS_PASSTHROUGH_CODEC_ILBC 0
#endif
as you can see i have disabled all the passthrough codecs except g729. but
still the g711 happens .
do i need to do something else for that? i would like to contribute to an
wiki on how to enable g729 on symbian pjsip
--
Regards
Shrouk Khan (Khan)
System Administrator / Telecommunication System Developer
Office: +354 4400807 (Reykjavik)
+44 2031370800 (London)
Mobile: +66 875049439 (Bangkok)
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