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have been fixed below:
- some UAs sends "telephone-event/8000/1" instead of
"telephone-event/8000", which caused SDP negotiation
to fail. Fixed in sdp_neg.c.
- codec name was (incorrectly) compared case-sensitively,
causing negotiation to fail. Fixed in sdp_neg.c.
- Also improved error reporting in SDP negotiation by
introducing few more error codes.
- Added Warning header in Not Acceptable response sent
by pjsip_inv_session when SDP negotiation fails.
- PJSUA-LIB will try to negotiate both SDPs before
sending 100 response.
- Fixed bug in iLBC codec when setting the mode to 30.
Also:
- Echo cancellation by default is disabled now since
it doesn't seem to work. Further investigation needed.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@738 74dad513-b988-da41-8d7b-12977e46ad98
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- in some condition, when outgoing call fails, call count
incorrectly decremented to -1
- introduce account priority in pjsua_acc_config, and
improve the account searching for incoming calls
- pjsua will hangup call after sending transfer/REFER request.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@737 74dad513-b988-da41-8d7b-12977e46ad98
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bridge, and resume as soon as frames are transmitted.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@731 74dad513-b988-da41-8d7b-12977e46ad98
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all the way up to PJSUA-API:
- standardized locking order: dialog then user agent, and dialog then PJSUA
- any threads that attempt to acquire mutexes in different order than
above MUST employ retry mechanism (for an example, see acquire_call() in
pjsua_call.c). This retry mechanism has also been used in the UA layer
(sip_ua_layer.c) since it needs to lock user agent layer first before
the dialog.
- introduced pjsip_dlg_try_inc_lock() and PJSUA_TRY_LOCK() to accomodate
above.
- pjsua tested on Quad Xeon with 4 threads and 200 cps, so far so good.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@729 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@719 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@691 74dad513-b988-da41-8d7b-12977e46ad98
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- API BREAK: pjsua_pres_create_uac() API CHANGED!! Added
options in the function, to allow creating SUBSCRIBE without
";id=" parameter in the Event header.
- the generic event publication in pjsip-simple/publish.[hc]
- split PIDF and X-PIDF body generation and parsing into
pjsip-simple/presence_body.c.
- allow NULL in module parameter in pjsip_endpt_add_capability()
- added "--publish" option in PJSUA.
- by default, PJSUA-LIB will not add ";id=" parameter in Event
header in the SUBSCRIBE request since lots of server and
user agents don't support this correctly.
- Set version to 0.5.7.6.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@685 74dad513-b988-da41-8d7b-12977e46ad98
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pjsua/pjsua-lib
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@683 74dad513-b988-da41-8d7b-12977e46ad98
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5 to use large resampling filter
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@659 74dad513-b988-da41-8d7b-12977e46ad98
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to reduce mic signal by division rather than zeroing the signal. Also moved out VAD and EC settings to config.h.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@658 74dad513-b988-da41-8d7b-12977e46ad98
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echo suppressor backend can be selected during runtime.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@653 74dad513-b988-da41-8d7b-12977e46ad98
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- Changed default sound backend in Windows to PortAudio
- Finalizing AEC settings on Windows:
- default tail is 256 msec
- write AEC configuration with "dc"
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@651 74dad513-b988-da41-8d7b-12977e46ad98
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sound_port, reduce DirectSound buffer from 32 to 16, and fixed ARM compilation for MSVC WinCE target.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@648 74dad513-b988-da41-8d7b-12977e46ad98
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disable AEC by default in pjsua-lib, and added ec-tail option in pjsua
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@643 74dad513-b988-da41-8d7b-12977e46ad98
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pjmedia and pjsua-lib.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@642 74dad513-b988-da41-8d7b-12977e46ad98
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- configurable default decoder mode (20 or 30),
- encoder mode follows the mode specified in SDP fmtp from
the remote's SDP,
- silence detector uses pjmedia's,
- PLC uses iLBC's PLC,
- perceptual enhancement (penh) is configurable via codec
param, as usual.
- iLBC mode is configurable in pjsua with --ilbc-mode option.
- Added packet lost simulation in pjmedia's UDP transport and
in pjsua (with --rx-drop-pct and --tx-drop-pct options).
- Increase default buffer count in DirectSound to 32 frames
to make it more resilient to CPU disruption.
- Specify and parse fmtp mode in SDP for codecs that need it.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@637 74dad513-b988-da41-8d7b-12977e46ad98
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conference bridge, also capability to use custom sound device in pjsua.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@632 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@629 74dad513-b988-da41-8d7b-12977e46ad98
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which means that some modules (evsub, invite) need to register their capabilities to the endpoint, (2) added command in pjsua to send arbitrary request
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@612 74dad513-b988-da41-8d7b-12977e46ad98
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header
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@582 74dad513-b988-da41-8d7b-12977e46ad98
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categories: (1) the TCP transport implementation itself (*.[hc]), (2) bug-fix in SIP transaction when using reliable transports, (3) support for TCP transport in PJSUA-LIB/PJSUA, and (4) changes in PJSIP-TEST to support TCP testing.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@563 74dad513-b988-da41-8d7b-12977e46ad98
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user_data from argument)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@541 74dad513-b988-da41-8d7b-12977e46ad98
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documentation
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@482 74dad513-b988-da41-8d7b-12977e46ad98
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high level API
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@476 74dad513-b988-da41-8d7b-12977e46ad98
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functionality, to allow using custom transports with streams
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@452 74dad513-b988-da41-8d7b-12977e46ad98
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- handle multiple frames in one packet
- split stream creation into two steps to allow customization
- PLC framework and implementation with G.711 and speex
- stream returns NO_FRAME correctly.
- added ptime argument in pjsua
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@438 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@360 74dad513-b988-da41-8d7b-12977e46ad98
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Changed pjmedia_endpt_create() API.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@350 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@343 74dad513-b988-da41-8d7b-12977e46ad98
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get_frame() when there is no listener on the port
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@318 74dad513-b988-da41-8d7b-12977e46ad98
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(some systems only have sound player)
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general bug
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and sampling rate. Also introduced sampling rate converter
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