- 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: https://svn.pjsip.org/repos/pjproject/trunk@780 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: https://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: https://svn.pjsip.org/repos/pjproject/trunk@742 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: https://svn.pjsip.org/repos/pjproject/trunk@737 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: https://svn.pjsip.org/repos/pjproject/trunk@729 74dad513-b988-da41-8d7b-12977e46ad98
- 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: https://svn.pjsip.org/repos/pjproject/trunk@685 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: https://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: https://svn.pjsip.org/repos/pjproject/trunk@637 74dad513-b988-da41-8d7b-12977e46ad98
Benny Prijono