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About: PJLIB, PJLIB-UTIL, PJSIP, and PJMEDIA are created by: Benny Prijono <bennylp pjsip.org>
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RELEASE NOTES
Version 0.5.8 2006/09/26
HIGHLIGHTS:
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Improved stability, quality, and
conformance (meaning: TONS of bug fixes).
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Autoconf based to support even more targets (new targets: FreeBSD, 64bit
Opteron or ItaniumII).
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Support for iLBC codec.
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Windows Mobile support.
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Support for PUBLISH.
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Support for call transfer with subscription suppression (norefersub).
-
Added tone generator (fixed or floating point).
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Experimental support for echo cancellation (suppressor or AEC).
INCOMPATIBILITIES WARNINGS:
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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.
-
Default PJLIB log size was increased from 1500 to 2000 bytes to accomodate
printing large SIP messages to the log. Beware when using PJ in embedded
applications since by default this buffer is allocated in the stack!
DETAILS:
GLOBAL:
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New autoconf based configuration (old configure was renamed to
configure-legacy).
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public header files should be ANSI and -pedantic clean, as well
as C++ friendly.
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fixes for 64bit targets (Opteron, Itanium II).
PJLIB:
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Added pj_thread_is_registered().
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Fixed mutex leaking in Linux: pj_mutex_destroy() will fail to destroy the mutex
if the mutex is currently being held.
-
Fix the local IP address resolution issue in all libraries.
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Added pool for allocating memory from the stack buffer.
PJLIB-UTIL:
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Change unescaping function in scanner and string.c to NOT unescape when '%' is
not followed by hex digits.
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Fix STUN client when handling unknown attributes.
PJMEDIA
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Added PJMEDIA_RTCP_IGNORE_FIRST_PACKETS macro to let application decides how
many first RTP packets to ignore when calculating jitter.
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Fixed rather loud clicking/tick noise when call is established. This was caused
by the PLC not properly cleared before it is reused for the next call.
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Bring speex codec up to date with their SVN trunk.
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Speex codec should work in FIXED_POINT mode when PJ_HAS_FLOATING_POINT is set
to zero.
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Added splitter/combiner port (splitcomb.c).
-
Added pjmedia_port_info_init() to fill in pjmedia_port_info structure.
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Added iLBC codec with full features (r637).
-
Added experimental echo canceller with backend can be set to simple suppressor
or Speex AEC.
-
Added pjsdp in pjmedia Makefile.
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Added multi-purpose tone generator in PJMEDIA (tonegen.[hc]).
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Fixed bug in resample port: put/get buffers are allocated with the wrong size
(thanks Danya).
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Added EOF callback in WAV/memory capture/playback ports.
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Fixed bug in media UDP transport: when transport is created with create(), no
IP address is given. This caused get_info() to return 0.0.0.0 as the transport
address.
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Removed G.711 Annex I PLC because of patent issues.
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Support for reading non-canonical WAV file in WAV file player port.
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Fixed race condition bug when initializing SDP parser on multi-processor
machine.
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Fixed SDP negotiation issues (r738).
PJSIP (Core, Ua, and SIMPLE)
-
Fixed bug in tel: URI unescaping code.
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Fixed bugs in client registration (sip_regc.c): when re-registration failed to
send the request, application callback is not called (r663).
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Fixed race condition and/or deadlock with SIP dialog, under stress testing.
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Added event publication (PUBLISH) support.
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Fixed authentication: by default, now PJSIP will not try to guess the
(Proxy-)Authorization header. It will just wait until it receives 401/407
before it sends the authorization info.
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Fixed authentication with qop (r696).
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Fixed bug in event subscription: for event subscriptions established by
non-SUBSCRIBE method (such as REFER), the method to refresh/unsubscribe
subscription would still be SUBSCRIBE, not the initial method to establish
subscription.
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SIP methods are now compared case-sensitively (r713).
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Fixed bug in client REGISTER: crashed when application calls
pjsip_regc_register()/pjsip_regc_send() while previous REGISTER request has not
been answered with final response. This occurs for example when
pjsip_regc_register() is called in quick successions (r714).
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Added pjsip_transport_register_type() API to register new transport type.
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Fixed race-condition/deadlock problems in the dialog/user agent layer all the
way up to PJSUA-API (r729).
PJSUA-LIB, PJSUA, and samples etc.
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Fixed minor bug in siprtp: in summary report, percentage of RX loss is
displayed in the TX section.
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Added --ptime and --no-vad option in pjsua.
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Added pjsua_wince application.
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Added Ctrl-C handler for pjsua Windows, and added average bps in media
statistic.
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Added call transfer with event suppression (norefersub) support.
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Added --duration option in PJSUA to limit the maximum duration of calls.
Go to the Download page to get
the version 0.5.8!
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