Age | Commit message (Collapse) | Author |
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1591 74dad513-b988-da41-8d7b-12977e46ad98
|
|
respond with a=inactive
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1562 74dad513-b988-da41-8d7b-12977e46ad98
|
|
- When UAS has sent answer in reliable 1xx, do not put SDP in 2xx
- Handle the case when UPDATE is challenged with 401/407
- Obsolete --service-route option in pjsua
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1561 74dad513-b988-da41-8d7b-12977e46ad98
|
|
the INVITE comes without SDP
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1553 74dad513-b988-da41-8d7b-12977e46ad98
|
|
Sipit21/Mon/12:30)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1549 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1538 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1533 74dad513-b988-da41-8d7b-12977e46ad98
|
|
call in PJSUA-LIB, with samples to send/receive DTMF with INFO in pjsua application
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1477 74dad513-b988-da41-8d7b-12977e46ad98
|
|
UPDATE
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1471 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1469 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1463 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1452 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1417 74dad513-b988-da41-8d7b-12977e46ad98
|
|
outgoing call (thanks Lemmel)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1406 74dad513-b988-da41-8d7b-12977e46ad98
|
|
does not have SDP, b) added on_create_offer() callback, c) handle some error cases
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1379 74dad513-b988-da41-8d7b-12977e46ad98
|
|
sending 487 (Role Conflict) response
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1291 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1266 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1264 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1242 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1134 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1133 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1112 74dad513-b988-da41-8d7b-12977e46ad98
|
|
in SDP (thanks Chris Hamilton)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1109 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1098 74dad513-b988-da41-8d7b-12977e46ad98
|
|
2007 in copyright notice in all sources
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@974 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@916 74dad513-b988-da41-8d7b-12977e46ad98
|
|
and minor fixes in PJSIP core implementation. Tested okay.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@881 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@875 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@863 74dad513-b988-da41-8d7b-12977e46ad98
|
|
when making outgoing call
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@847 74dad513-b988-da41-8d7b-12977e46ad98
|
|
media problems
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@831 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@813 74dad513-b988-da41-8d7b-12977e46ad98
|
|
- Added support for SIP Replaces extension (RFC 3891)
- Added pjsua_call_xfer_replaces() to perform attended call
transfer.
- PJSUA checks and process Replaces header in incoming calls
- Added pjsip_ua_find_dialog() API.
- Added pjsip_endpt_has_capability() API.
- Added pjsip_endpt_send_response2() API.
- etc.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@797 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@782 74dad513-b988-da41-8d7b-12977e46ad98
|
|
- added callback to report call transfer progress
- changed the call transfer request callback name in pjsua
- added "--norefersub" option in pjsua.
- fixed bug when call transfer is done more than once in
the same dialog (dialog usage can not be added)
Also removed 7xx status from the SIP status codes.
And added pjsip_parse_status_line() to parse sipfrag.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@780 74dad513-b988-da41-8d7b-12977e46ad98
|
|
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
|
|
- 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
|
|
locking algorithm:
- Fixed crash in PJSUA-API when initiating client subscription
- Fixed another crash in PJSUA-API when hanging-up call
Also fixed SDP negotiator:
- add a=inactive when rejecting media line
Also increase maximum log size from 1500 to 2000 since some
SIP packet is quite large. A little bit of Warning:
** THIS MAY AFFECT APPLICATION'S STACK USAGE **
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@734 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@733 74dad513-b988-da41-8d7b-12977e46ad98
|
|
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
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@719 74dad513-b988-da41-8d7b-12977e46ad98
|
|
properly register all supported SIP method, accepted content type, and supported extensions to endpoint.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@688 74dad513-b988-da41-8d7b-12977e46ad98
|
|
statistic.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@660 74dad513-b988-da41-8d7b-12977e46ad98
|
|
- 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
|
|
- 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
|
|
chars (from 16), and check all those sprintf's especially the ones with "%p" format.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@635 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@629 74dad513-b988-da41-8d7b-12977e46ad98
|
|
header
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@611 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@597 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@582 74dad513-b988-da41-8d7b-12977e46ad98
|