If you are using PJSIP 1.3
Edit config_site.h
#if defined(PJ_SYMBIAN) || PJ_SYMBIAN==1
# define PJMEDIA_AUDIO_DEV_HAS_SYMB_APS 1
# define PJMEDIA_AUDIO_DEV_HAS_SYMB_MDA 0
#endif
// ...
#define PJMEDIA_CONF_USE_SWITCH_BOARD 1
// Below, enable all codecs you need
//...
//Enable passthrough codecs
#define PJMEDIA_HAS_PASSTHROUGH_CODECS 1
// Disable all passthrough codecs except PCMA and PCMU
#define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMU 1
#define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMA 1
#define PJMEDIA_HAS_PASSTHROUGH_CODEC_AMR 0
#define PJMEDIA_HAS_PASSTHROUGH_CODEC_G729 0
#define PJMEDIA_HAS_PASSTHROUGH_CODEC_ILBC 0
Edit you mmp project file:
#define SND_HAS_APS 1
#define SND_HAS_VAS 0
#define SND_HAS_MDA 0
Specify application UID in a global variable named APP_UID whose base type TPtrC, e.g:
TPtrC APP_UID = _L("2000521C");
Link the application to APS library, i.e:
LIBRARY APSSession2.lib
Add MultimediaDD capability to your application, i.e. in application MMP:
CAPABILITY MultimediaDD ...
Install APS Server apsserver2.sisx to device, the installer comes with the APS SDK package i.e in folder InstallToDevice/AudioProxyServer243. You will need to do this only once for each device.
The complete step-by-step process is detailed in:
http://trac.pjsip.org/repos/wiki/APS
and
http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct
[]'s
Rafael Donato.
ltquang at tma.com.vn wrote:
> 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
>
>
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