Hi benny:
Thanks for reply, your explain are clear,now I understand it,thanks very much.
Actually,my real problem is,the audio quality unsatisfactory when I run app in mobile(Symbian VAS,G.729),so I doubt it's problem of VAS,because the quality is accepted when I use VAS AMR.
Today I found the real reason is jitter buffer! The quality improved when Iincrease the value of pjsua_media_config.jb_xxx, since my app run on GPRS/EDGE,I think the RTT is bigger than ADSL/WLAN,so the jb need bigger delay,is it correct? and how to set appropriate value of pjsua_media_config.jb_xxx according to RTT? Thanks again.
Best regards
2009-11-12
mylen
In non-"Direct" setup, pjsip has sound device, conference bridge, and
software codecs (like Speex. GSM, etc.). So no, we don't support G.729
in non-Direct setup since we don't have G.729 software codec.
The xxx-Direct feature was developed to support hardware codecs that
sound devices have, such as G.729, AMR NB, iLBC, etc in VAS/APS. Since
the audio from/to the sound device is already in encoded format, we
don't need to encode/decode the audio with the usual software codecs
anymore, that's why we invented passthough codecs, which basically
just provide the codec info for SDP, and doesn't do any real
encoding/decoding. Also because audio is encoded, we can't use the
conference bridge anymore, since we can't mix encoded audio. Hence the
switchboard.
But even when xxx-Direct is used, you can still enable/use software
codecs if you want to, such as Speex and GSM, since these codecs are
not supported by the devices. When this codec is selected for the
call, PJSUA-LIB will take care that VAS/APS is opened in the correct
mode (i.e. PCM) to handle this. But you still loose the conferencing
feature since the switchboard instead of conference bridge is used
(the selection between switchboard and confbridge is done at
compile-time rather than run-time).
Hope this helps. Let me know if you still have any more doubts.
Cheers
Benny
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