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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20101014/b0171539/attachment.html>