- special treatment in ICE media transport, since Session Timer refresh when sending re-INVITE doesn't call transport_encode_sdp(), causing the ICE media transport to mistakenly think that it's role is UAS hence it will disable current ICE session
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@2896 74dad513-b988-da41-8d7b-12977e46ad98
- Removed self detach() in destroy() in transport SRTP.
- Added few more assertion checks in transport SRTP.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@2885 74dad513-b988-da41-8d7b-12977e46ad98
- Assertion in pjmedia tests's MIPS test with iLBC codec due to ptime mismatch
- added compile time macro in test.h to enable/disable specific tests
- also added SVN keyword expansion attributes to the source files
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@2882 74dad513-b988-da41-8d7b-12977e46ad98
WSOLA improvements:
- Introduce fade-out and fade-in effect
- Limit the number of continuous synthetic samples (only take effect when fading is used)
- Export many settings as macros:
- PJMEDIA_WSOLA_DELAY_MSEC (was HANNING_PTIME)
- PJMEDIA_WSOLA_TEMPLATE_LENGTH_MSEC (was TEMPLATE_PTIME)
- PJMEDIA_WSOLA_MAX_EXPAND_MSEC
PLC:
- added compile time macro PJMEDIA_WSOLA_PLC_NO_FADING to disable fading (default enabled)
Stream:
- fixed bug when stream is not PLC-ing subsequent packet loss (only the first)
- also add maximum PLC limit just as precaution if PLC doesn't limit number of synthetic frames
- unrelated: fixed warning about unused send_keep_alive() function
Delaybuf:
- modified to NOT use fading in WSOLA since we don't expect it to generate many continuous synthetic frames
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@2850 74dad513-b988-da41-8d7b-12977e46ad98
- This ticket allows the same loop media transport instance to be attached to more than one streams, and allow application to control which stream(s) receives the reflected packets.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@2845 74dad513-b988-da41-8d7b-12977e46ad98
- Added support for Nokia VAS 2.0.
- Fixed wrong value assigned to last downstream state var in downstream callback.
- Minor fix in config_site_sample.h related to VAS Direct setting.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@2833 74dad513-b988-da41-8d7b-12977e46ad98
- Added a new API pjmedia_codec_passthrough_init2().
- Updated the initialization steps of passthrough codec in pjsua_media.c, to configure the codecs (of passthrough codec) to be enabled based on audio device extended/encoded formats.
- Minor update: added passthrough.h into pjmedia_codec.vcproj.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@2825 74dad513-b988-da41-8d7b-12977e46ad98
- Added new audio device VAS for Symbian platform.
- Updated symsndtest to use the latest audio device framework.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@2821 74dad513-b988-da41-8d7b-12977e46ad98
- Fixed stream bug in generating RTP header for keep-alive packet with empty RTP.
- Updated stream to skip suspending the VAD when PJMEDIA_STREAM_VAD_SUSPEND_MSEC is set to 0.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@2761 74dad513-b988-da41-8d7b-12977e46ad98
- Fixed bug in conf bridge create_on_port() (bug of r2747).
- Updated sound port to instantiate sync port only when audio format is PCM.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@2748 74dad513-b988-da41-8d7b-12977e46ad98
- when --disable-sound is specified, no sound factory will be created
- modified audiodev.c to not return error when there's no device found
- compilation error in pa_dev.c when it's disabled and include path is not set
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@2740 74dad513-b988-da41-8d7b-12977e46ad98
- Added calls to delay buf destructor in conference.c and echo_common.c.
- Moved mutex creation to the end of pjmedia_delay_buf_create().
- Deprecated pjmedia_conf_add_passive_port().
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@2728 74dad513-b988-da41-8d7b-12977e46ad98
- #793: AMR encoder should regard 'mode-set' param specified by remote decoder.
- #831: Automatically switch to TCP transport when sending large request
- #832: Support for outbound proxy setting without using Route header
- #849: Modify conference audio switch behavior in connecting ports.
- #850: Remove 'Require=replaces' param in 'Refer-To' header (in call transfer with replaces).
- #851: Support for regular nomination in ICE
- #852: --ip-addr support for IPv6 for media transport in pjsua
- #854: Adding SOFTWARE attribute in all outgoing requests may cause compatibility problem with older STUN server (thanks Alexei Kuznetsov for the report)
- #855: Bug in digit map frequencies for DTMF digits (thanks FCCH for the report)
- #856: Put back the ICE candidate priority values according to the default values in the draft-mmusic-ice
- #857: Support for ICE keep-alive with Binding indication
- #858: Do not authenticate STUN 438 response
- #859: AMR-WB format param in the SDP is not negotiated correctly.
- #867: Return error instead of asserting when PJSUA-LIB fails to open log file
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@2724 74dad513-b988-da41-8d7b-12977e46ad98
- Updated APS to supports for sampling rate 16kHz and stereo (using resample).
- Fixed buffer len calculations in Symbian MDA.
- Disabled multichannel support on Symbian MDA (since its input device doesn't support for stereo?).
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@2677 74dad513-b988-da41-8d7b-12977e46ad98
Nanang Izzuddin
Benny Prijono