I retrieved the latest from the Subversion database and have built an image that almost works in the CE 5.0 environment. I am testing against SJphone running on the PC. As I previously mentioned, I had to change the definition of VSNPRINTF in pa_debugprint.c to get things to compile, and changed a few things in pjsua_wince.cpp: the value of SIP_DST_URI, disabled STUN and disabled ICE. I am testing on a local network, so STUN and ICE would seem to be unnecessary. I can establish connection in either direction, either from SJphone to wince_demos or visa versa. The audio from the PC to the WinCE computer seems fine. Clear and very little delay; much less delay than with Microsoft's CE VOIP demo (which seemed excessive). Here is the problem: the audio from the CE box toward the PC is choppy and eventually drops down to a few clicks. At this point, few RTP packets are being sent toward the PC (as indicated by WireShark). Those few remaining packets being sent by wince_demos all have the "Mark" bit set. I have my options in config_site.h set as follows: # define PJ_HAS_FLOATING_POINT 1 # define PJMEDIA_HAS_G722_CODEC 0 # define PJMEDIA_HAS_G711_PLC 0 # define PJMEDIA_HAS_L16_CODEC 0 # define PJMEDIA_HAS_GSM_CODEC 1 # define PJMEDIA_HAS_ILBC_CODEC 0 # define PJMEDIA_HAS_SPEEX_CODEC 0 # define PJMEDIA_HAS_SPEEX_AEC 0 # undef PJMEDIA_RESAMPLE_IMP # define PJMEDIA_RESAMPLE_IMP PJMEDIA_RESAMPLE_LIBRESAMPLE # define PJMEDIA_WSOLA_IMP PJMEDIA_WSOLA_IMP_NULL # define PJMEDIA_HAS_SRTP 0 WireShark indicates that GSM codecs are used in both directions. Not sure where to look next. Suggestions? -- Jack Bonn <> Software Design Labs, Inc. jack.bonn at swlabs.com (847)526-1337 Dyslexics untie.
