Does anyone success on enabling APS-Direct on Symbian S60 3rd MR? How can I do to enable it? I tried to set #define PJ_CONFIG_NOKIA_APS_DIRECT 1 #define PJMEDIA_AUDIO_DEV_HAS_SYMB_APS 0 #define PJMEDIA_AUDIO_DEV_HAS_SYMB_MDA 0 #define PJMEDIA_AUDIO_DEV_HAS_SYMB_VAS 0 in file config_site_sample.h but when I make call, my app crashes right after receiving SIP 200 OK. Looking into the log file, PJSIP said: 09/09/15_14:16:55,947500 APS initialized 09/09/15_14:16:55,954125 Adjusting quality to 5 for uwb 09/09/15_14:16:55,960125 Creating conference bridge with 12 ports 09/09/15_14:16:55,967125 Sound device successfully created for port 0 09/09/15_14:16:55,975125 WARNING: no real random source present! 09/09/15_14:16:55,981125 Module "mod-evsub" registered 09/09/15_14:16:55,988375 Module "mod-presence" registered 09/09/15_14:16:55,994375 Event pkg "presence" registered by mod-presence 09/09/15_14:16:56,002000 Module "mod-refer" registered 09/09/15_14:16:56,014000 Event pkg "refer" registered by mod-refer 09/09/15_14:16:56,022000 Module "mod-pjsua-pres" registered 09/09/15_14:16:56,028000 Module "mod-pjsua-im" registered 09/09/15_14:16:56,034000 Module "mod-pjsua-options" registered 09/09/15_14:16:56,042000 No SIP worker threads created 09/09/15_14:16:56,048000 pjsua version 1.4 for symbian initialized 09/09/15_14:16:56,054000 SIP UDP socket reachable at 192.168.100.96:50600 09/09/15_14:16:56,060000 Error setting SO_RCVBUF: Invalid operation (PJ_EINVALIDOP) [70013] 09/09/15_14:16:56,066000 Error setting SO_SNDBUF: Invalid operation (PJ_EINVALIDOP) [70013] 09/09/15_14:16:56,086125 SIP UDP transport started, published address is 192.168.100.96:50600 09/09/15_14:16:56,093125 Account <sip:192.168.100.96:50600> added with id 0 09/09/15_14:16:56,102750 RTP socket reachable at 192.168.100.96:4000 09/09/15_14:16:56,109375 RTCP socket reachable at 192.168.100.96:4001 09/09/15_14:16:56,126000 Module "mod-siprtp-log" registered 09/09/15_14:16:56,230750 Account <sip:38409823 at as.fibertel.dk> added with id 1 09/09/15_14:16:56,237750 Request msg REGISTER/cseq=53855 (tdta0x772de0) created. 09/09/15_14:16:56,243750 Transaction created for Request msg REGISTER/cseq=53856 (tdta0x772de0) 09/09/15_14:16:56,250000 Sending Request msg REGISTER/cseq=53856 (tdta0x772de0) in state Null 09/09/15_14:16:56,256000 Target '77.233.244.20:0' type=Unspecified resolved to '77.233.244.20:5060' type=UDP (UDP transport) ... 09/09/15_14:16:59,498375 RX 970 bytes Response msg 200/INVITE/cseq=43 (rdata0x71b304) from UDP 77.233.244.20:5060: SIP/2.0 200 OK From: <sip:38409823@xxxxxxxxxxxxxx>;tag=UVwOCdt4cctUIq60P-UTH2U1uK1NhJaB To: <sip:38409820 at as.fibertel.dk>;tag=cf2e94d-13c4-4aaf85bd-fc440475-fc72272 Call-ID: wvqkq5SCYMe1a7O0ALAbQfWep8Z3Gs77 CSeq: 43 INVITE Remote-Party-ID: "tmatest tmatest" <sip:9820 at 77.233.242.41;user=phone>;screen=yes;party=called;privacy=off;id-type=subscriber Accept: multipart/mixed,application/media_control+xml,application/sdp Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Via: SIP/2.0/UDP 192.168.100.96:50600;received=87.48.133.162;rport=15427;branch=z9hG4bKPjLlTjd41NHNFuBDYAM9sZx254OZ52yGFJ Contact: <sip:38409820 at 77.233.244.20:5060;transport=UDP> Content-Type: application/sdp Content-Length: 240 v=0 o=BroadWorks 6074 1 IN IP4 77.233.244.20 s=- c=IN IP4 77.233.244.20 t=0 0 m=audio 23736 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --end msg-- 09/09/15_14:16:59,515375 Incoming Response msg 200/INVITE/cseq=43 (rdata0x71b304) in state Proceeding 09/09/15_14:16:59,522375 State changed from Proceeding to Terminated, event=RX_MSG 09/09/15_14:16:59,528375 Received Response msg 200/INVITE/cseq=43 (rdata0x71b304) 09/09/15_14:16:59,534375 Route-set updated 09/09/15_14:16:59,540375 Route-set frozen 09/09/15_14:16:59,548375 Transaction tsx0x71a32c state changed to Terminated 09/09/15_14:16:59,562375 Got SDP answer in Response msg 200/INVITE/cseq=43 (rdata0x71b304) 09/09/15_14:16:59,568375 SDP negotiation done, status=0 09/09/15_14:16:59,574375 Call 0: remote NAT type is 0 (Unknown) 09/09/15_14:16:59,584125 pjmedia_rtp_session_init: ses=0x77e2e4, default_pt=8, ssrc=0x1b5d8f3 09/09/15_14:16:59,590125 pjmedia_rtp_session_init: ses=0x77e908, default_pt=8, ssrc=0x1b5d8f3 09/09/15_14:16:59,596125 Stream strm0x77cce4 created 09/09/15_14:16:59,602125 Encoder stream started 09/09/15_14:16:59,608125 Decoder stream started 09/09/15_14:16:59,614125 Media updates, stream #0: PCMA (sendrecv) 09/09/15_14:16:59,622125 Opening sound device ALAW at 8000/1/60ms 09/09/15_14:17:00,152625 Port 1 (sip:38409820 at as.fibertel.dk) transmitting to port 0 (S60 APS) then PJSIP crashes Please help me
