Age | Commit message (Collapse) | Author |
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@862 74dad513-b988-da41-8d7b-12977e46ad98
|
|
options, and pjsua integration. The TLS support should work in both client and server mode.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@861 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@858 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@854 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@853 74dad513-b988-da41-8d7b-12977e46ad98
|
|
request but somehow connection gets closed by server after the request is sent
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@852 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@849 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
|
|
but rather only contact that was previously sent in the registration. In addition, added function pjsip_regc_unregister_all() to unregister all contacts
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@843 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@841 74dad513-b988-da41-8d7b-12977e46ad98
|
|
after pjsua_destroy() is called
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@839 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@834 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@833 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@824 74dad513-b988-da41-8d7b-12977e46ad98
|
|
to work much better now and take less CPU, so I increased
default tail length in PJSUA to 800ms.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@823 74dad513-b988-da41-8d7b-12977e46ad98
|
|
and also consider sound device latency when applying EC. It
sounds like working although it still doesn't perfectly cancel
the echo.
EC is now by default enabled in PJSUA.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@822 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@818 74dad513-b988-da41-8d7b-12977e46ad98
|
|
pj_atexit(). Also fixed handle leaks in SIP transaction layer and SIP endpoint.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@815 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@813 74dad513-b988-da41-8d7b-12977e46ad98
|
|
because this caused pasound.c to autodetect default device
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@812 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@811 74dad513-b988-da41-8d7b-12977e46ad98
|
|
PJ_IOQUEUE_MAX_HANDLES mutex to leak during program exits
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@810 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@808 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@801 74dad513-b988-da41-8d7b-12977e46ad98
|
|
to the REGISTER request. This solves the problem where headers
registered in the initial REGISTER request (such as User-Agent
header) are not sent in subsequent reregistration request.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@799 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@794 74dad513-b988-da41-8d7b-12977e46ad98
|
|
proxies will receive multiple Via headers.
Thanks Aldo <acampi at deis.unibo.it>.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@792 74dad513-b988-da41-8d7b-12977e46ad98
|
|
and quality to comply with LAME, also changed the pjsua_recorder_create() parameter to allow specifying mp3 options in one of the parameter
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@785 74dad513-b988-da41-8d7b-12977e46ad98
|
|
"--auto-rec" option in pjsua to record voice conversion. The "--rec-file" option will record to either .WAV or .MP3 depending on the file extension.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@783 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@782 74dad513-b988-da41-8d7b-12977e46ad98
|
|
1024 from 512
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@781 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
|
|
OF THE STATUS!!!
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@777 74dad513-b988-da41-8d7b-12977e46ad98
|
|
immediate error
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@776 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@769 74dad513-b988-da41-8d7b-12977e46ad98
|
|
sampling rate than the clock rate configuration, resampling port needs to be created.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@765 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@763 74dad513-b988-da41-8d7b-12977e46ad98
|
|
about the failure (so that ACK is transmitted first before next INVITE is sent).
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@756 74dad513-b988-da41-8d7b-12977e46ad98
|
|
- added DNS asynchronous/caching resolver engine in
PJLIB-UTIL (resolver.[hc])
- modified SIP resolver (sip_resolve.c) to properly
perform DNS SRV/A resolution when DNS resolution is
enabled.
- added dns_test.c in PJSIP-TEST for testing the SIP
resolver.
- added nameserver configuration in PJSUA-LIB
- added "--nameserver" option in PJSUA.
- updated project/Makefiles and doxygen documentation.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@753 74dad513-b988-da41-8d7b-12977e46ad98
|
|
This option can be used for example to select the IP
interface of SIP/RTP/RTCP transports, or to specify the
public IP address of NAT/router in case port forwarding is
used.
For SIP transports, this feature works for both UDP and
TCP transports.
Changes:
- added public_ip field in pjsua_transport_config, and
change SIP and media transport creation to consider this
option.
- added --ip-addr option in pjsua
- added pjsip_tcp_transport_start2() which allows
specifying alternate TCP published address when creating
TCP transports.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@742 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
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@736 74dad513-b988-da41-8d7b-12977e46ad98
|
|
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@735 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
|
|
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
|
|
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
|