- Timestamp:
- Jun 2, 2008 6:30:15 PM (16 years ago)
- Location:
- pjproject/trunk
- Files:
-
- 1 added
- 7 edited
Legend:
- Unmodified
- Added
- Removed
-
pjproject/trunk/build.symbian/bld.inf
r1978 r1979 18 18 pjsip_ua.mmp 19 19 pjsua_lib.mmp 20 libsrtp.mmp 20 21 21 22 /* Codecs */ -
pjproject/trunk/build.symbian/pjlib_util.mmp
r1765 r1979 30 30 // PJLIB-UTIL files 31 31 // 32 SOURCE base64.c 32 33 SOURCE crc32.c 33 34 SOURCE dns.c -
pjproject/trunk/build.symbian/pjmedia.mmp
r1965 r1979 70 70 SOURCE transport_ice.c 71 71 SOURCE transport_udp.c 72 SOURCE transport_srtp.c 72 73 SOURCE wav_player.c 73 74 SOURCE wav_playlist.c … … 93 94 SYSTEMINCLUDE ..\pjlib-util\include 94 95 SYSTEMINCLUDE ..\pjnath\include 96 SYSTEMINCLUDE ..\third_party\srtp\include 97 SYSTEMINCLUDE ..\third_party\srtp\crypto\include 98 SYSTEMINCLUDE ..\third_party\build\srtp 95 99 96 100 SYSTEMINCLUDE \epoc32\include -
pjproject/trunk/build.symbian/symbian_ua.mmp
r1965 r1979 32 32 LIBRARY pjsip_simple.lib pjsip.lib pjsdp.lib pjmedia.lib 33 33 LIBRARY pjnath.lib pjlib_util.lib pjlib.lib 34 LIBRARY symbian_audio.lib 35 LIBRARY libgsmcodec.lib 34 LIBRARY symbian_audio.lib libsrtp.lib 35 LIBRARY libgsmcodec.lib libspeexcodec.lib 36 36 #else 37 37 STATICLIBRARY pjsua_lib.lib pjsip_ua.lib 38 38 STATICLIBRARY pjsip_simple.lib pjsip.lib pjsdp.lib pjmedia.lib 39 39 STATICLIBRARY pjnath.lib pjlib_util.lib pjlib.lib 40 STATICLIBRARY symbian_audio.lib 40 STATICLIBRARY symbian_audio.lib libsrtp.lib 41 41 STATICLIBRARY libgsmcodec.lib libspeexcodec.lib 42 42 #endif -
pjproject/trunk/pjlib/include/pj/config_site_sample.h
r1971 r1979 46 46 47 47 /* SRTP has not been ported to Symbian yet */ 48 # define PJMEDIA_HAS_SRTP 048 # define PJMEDIA_HAS_SRTP 1 49 49 50 50 /* Disable these */ -
pjproject/trunk/pjsip-apps/src/symbian_ua/ua.cpp
r1965 r1979 78 78 #define USE_ICE 1 79 79 80 // 81 // Use SRTP? 82 // 83 #define USE_SRTP PJSUA_DEFAULT_USE_SRTP 80 84 81 85 // … … 290 294 cfg.max_calls = 2; 291 295 cfg.thread_cnt = 0; // Disable threading on Symbian 296 cfg.use_srtp = USE_SRTP; 297 cfg.srtp_secure_signaling = 0; 298 292 299 cfg.cb.on_incoming_call = &on_incoming_call; 293 300 cfg.cb.on_call_media_state = &on_call_media_state; -
pjproject/trunk/pjsip-apps/src/symbian_ua_gui/src/symbian_ua.cpp
r1973 r1979 27 27 #define SIP_PORT 5060 28 28 #define USE_ICE 0 29 #define USE_SRTP PJSUA_DEFAULT_USE_SRTP 29 30 30 31 static RSocketServ aSocketServer; … … 303 304 cfg.max_calls = 2; 304 305 cfg.thread_cnt = 0; // Disable threading on Symbian 305 //cfg.use_srtp = 0;306 //cfg.srtp_secure_signaling = 0;306 cfg.use_srtp = USE_SRTP; 307 cfg.srtp_secure_signaling = 0; 307 308 308 309 cfg.cb.on_incoming_call = &on_incoming_call;
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