PJSIP version 0.5.10.3

Released on 2007-04-11T15:53:40Z

Tickes/Bugs Fixed or Pending on This Release## common

For more details about the ticket list, please see Release Notes

PJSIP version 0.5.10.2

Released on 2007-03-13T23:38:58Z

Release Notes## common

For more details about the ticket list, please see Release Notes

PJSIP version 0.5.10.1

Released on 2007-02-20T00:32:22Z
Tickets fixed on this release:

For more details about the ticket list, please see Release Notes

PJSIP version 0.5.10

Released on 2007-02-19T04:35:11Z
Release Notes for 0.5.10

ChangesWhen updating to version 0.5.10, please be aware of the following changes:

• Since Visual Studio 8/2005 support is now included in the distribution, you will need to delete your VS 2005 project files and use the one that are with the tarball instead.
• There is an API change with pjsip_tpmgr_acquire_transport() function. This should not break any applications unless you are dealing with the very low level SIP transport.

Other than those, things should be compatible with 0.5.9 version.

New FeaturesThis release contains the following new features.

Python Wrapper for PJSUA-API

An experimental implementation of Python module to access PJSUA-API is available in pjsip-apps/src/py_pjsua directory. The [/pjsip/docs/html/group__PJSUA__LIB.htm PJSUA-API Documentation] has been updated with the Python information (e.g. every structure and function are accompanied by the corresponding Python information). Also have a look at the mini [/trac/wiki/Python_SIP_Tutorial SIP/Media Python Tutorial].

Visual Studio 2005 Support

After long overdue, this release would now support Visual Studio 2005 IDE (ticket #72). Please find new workspace for VS6/VS8 in pjproject root directory. The VS6 will continue to be supported.

TLS Support

An experimental TLS support based on OpenSSL library is implemented (see ticket #3).

Currently the TLS transport still blocks in SSL_connect() and SSL_accept() (although the TCP connect/accept procedure is non-blocking). An enhancement for this is planned (see ticket #63).

Explicit Transport Binding

Previously, transport is selected automatically from transport hash table to send outgoing requests/responses, based on the rules in various RFCs. This works okay for typical user agent application, but some other types of applications (such as proxies or test agents) want to have more control over which transport to be used to send messages.

This release adds functions to control transport to be used by dialogs, transactions, client registration, and low-level API to set which transport to be used when sending messages:

• pjsip_dlg_set_transport()
• pjsip_tsx_set_transport()
• pjsip_regc_set_transport()
• pjsip_tx_data_set_transport()
• pjsua_acc_set_transport()

See ticket #50 for more info.

DTMF Callback Support

Previously PJSUA-API lacks capability to report incoming DTMF digits to application. This feature has been implemented (see ticket #48).

More NAT Friendly

• The RTP stream now can be configured to periodically transmit RTP packets during silence period, to maintain NAT binding and to prevent intelligent server from terminating the call because of inactivity (see ticket #56).
• Disable VAD for the first few hundreds of milliseconds after the stream is started to open local and remote NAT bindings and to let remote knows about the source address of the RTP packets (see ticket #53).

More Robust SDP Parsing

• Allow parsing of SDP message containing invalid whitespaces (ticket #64).
• Allow empty format list in pjmedia_sdp_validate() when media port is zero (ticket #75).
  Thanks Briac Hardouin.

Faster and Better G.711 ALaw/ULaw Codec

Toni Rutar @aufbix.org has contributed a table based implementation of G.711 ALaw/ULaw codec which is much faster than existing one, and it doesn't have codec impairment of existing PCMA implementation. The new implementation will be used by default, but the old implementation can still be used for small footprint devices (see ticket #26)

WAV File Playlist

David Clark @tx.rr.com has contributed a WAV file playlist media port (to play multiple WAV files in a play list). Many thanks. (see ticket #67). Alain Totouom uses the playlist and found/fixed some problems :D (see ticket #86).

Misc

Multiple WAV Files in pjsua

pjsua now can accepts multiple --play-file to load multiple WAV files (ticket #77)

More Robust Timeout Handling in Client Registration

Handle the situation where server crashes after sending 100/Trying to REGISTER, with a timeout timer (ticket #99 and #89).

Play tone option in pjsua

Added --play-tone option in pjsua (ticket #62)

Playing Tone in a Loop

Added PJMEDIA_TONEGEN_LOOP option in tone generator/pjmedia to play tone continuously (ticket #62).

QoS/ToS Abstraction in PJLIB

Added socket constants abstraction to set up socket TOS/QoS in PJLIB (ticket #83, thanks Roman Puls for the suggestion and info).

New PJSUA APIs

• Individual conference port audio level adjustment (ticket #51)
• API to retrieve media port associated with file player/writer (ticket #52)
• Playlist creation APIs (ticket #67).

Other

• Add socket handles to UDP media transport info (ticket #34)
• One can now feed a simple text file containing commands to pjsua via shell redirection (thanks Steven Halerman).
• Added PJSUA_MAX_PLAYER and PJSUA_MAX_RECORDER constants in PJSUA-API (ticket #102, thanks David Clark for the report)

Bug Fixes[/trac/query?status=closed&milestone=release-0.5.10 Many bugs have been fixed]. Thanks for everybody who have reported them!

PJLIB/PJLIB-UTIL bug fixes (major bugs are in bold font):

• The select ioqueue ignores the flag specified in pj_ioqueue_recv()/recvfrom() (ticket #49)
• When host doesn't have any NIC interfaces, it should return 127.0.0.1 instead of error (ticket #71)
• Broken compilation with gcc-2.9x or older (ticket #66).
• Th pj_gethostbyname() function should return the correct error code when fails (ticket #80)
• Win32 now uses native Win32 API for file I/O to get rid of FOPEN_MAX limit (ticket #103, thanks David Clark for the report)
• Write outside buffer in PJLIB's win32 error string (ticket #106)

PJSIP bug fixes:

• Authentication loops forever when (malfunction) server keeps rejecting request with stale=true (ticket #25)
• TCP (and TLS) transport is incorrectly removed from hash table when TCP connection is closed by remote and new connection is created to replace it (ticket #42)
• Accept, Allow, Supported, and other array of tokens header should allow empty content (ticket #45)
• pjsip_regc_unregister() now will unregister local contact instead of all contacts (with "*") (ticket #35)
• Unable to send registration in registration callback (ticket #34)
• Registration did not respond to 401/407 challenge on unregistration after it has been marked for deletion (ticket #73)
• Registration crashes when NIC is plugged-off during registration refresh(ticket #81, thanks Richard Allister for reporting the bug)
• Bug with URI copying in strict routing handling causing duplicate "<<" characters in Route header (ticket #97, thanks Ryder Rishel for investigating and fixing the problem).
• pjsip_ctype_hdr_clone() doesn't duplicate media subtype (ticket #104, thanks Igor S for the report!)

PJMEDIA bug fixes:

• Crash in echo suppressor on Windows Mobile (ticket #43)
• A wrong "Invalid RTP version" message is printed in some cases (ticket #47)
• Wrong PortAudio error mapping causes PortAudio to malfunction (ticket #55)
• Frame timestamp was not propagated correctly by the conference bridge (minor, ticket #70).
• RTCP discard statistic should be increased when invalid RTP packet is received (ticket #37)
• More stable audio on Windows Mobile (ticket #74, thanks ChenHuang for the report and solution)
• Allow empty format list in pjmedia_sdp_validate() when media port is zero (ticket #75, thanks Briac Hardouin for the report and solution).
• Bug in PortAudio on MacOSX 10.4 causing no audio input when device is opened with non-native clock rate (ticket #76, thanks Norman Franke for the report and solution)
• Bug in conference bridge/resampling causing noisy audio with some clock rate settings (ticket #78, thanks Norman Franke for the report)
• Bug in conference bridge in level adjustment calculation (ticket #94, thanks frank wiersma for the fix).
• Fixed linking errors when codecs are disabled from configure script (ticket #82, thanks Thomas Reitmayr)
• Audio saturation when level is applied to conference (ticket #109)

PJSUA-API bug fixes:

• Cannot re-initialize PJSUA with pjsua_create() after pjsua_destroy() (ticket #29)
• Misleading error message in PJSUA when outgoing call fails because of invalid URI (ticket #38)
• Bug when account is deleted and new account is created, previous session is not completely cleared (ticket #68)
• pjsua_transport_close() doesn't properly close transports (especially TLS) (ticket #84, thanks Alain Totouom)
• Error when binding SIP transport to specific interface (ticket #98, thanks Hoi-Ho Chan for reporting).
• Error when specifying port zero for SIP transports (ticket #101).

For more details about the ticket list, please see Release Notes

PJSIP version 0.5.9

Released on 2006-11-27T00:00:00Z
Version 0.5.9 release contains the following fixes/improvements.

New Features* Attended call transfer implementation (Replaces support, RFC 3515).

• Major addition to support DNS SRV resolution in PJSIP:
  • Implemented generic asynchronous DNS resolution engine in PJLIB-UTIL: [http://www.pjsip.org/pjlib-util/docs/html/group__PJ__DNS__RESOLVER.htm]
  • Implemented SRV support in PJSIP: [http://www.pjsip.org/pjsip/docs/html/group__PJSIP__RESOLVE.htm]
• MP3 file recording [contributed by Toni Rutar]

Improvements* the echo canceller seems to work and it's now enabled by default on PJSUA-API

• Speex source codes updated to their latest SVN version (version later than 1.2beta1)
• added pj_shutdown() and pj_atexit() to release internal handles allocated by libraries (important for Symbian)
• Add pjsip_regc_add_headers() APi to set headers to be added 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.
• Improve call transfer handling in PJSUA-API (not to disconnect call until call transfer completes)
• Improvements in audio device support:
  • Updated PortAudio to latest version for Mac (OSX 10.4 compatibility)
  • PJSUA-API will try to use other sample rates when it fails to open the device
  • Better support for opening different sound devices.
• Added support for specifying IP address in PJSUA-LIB/pjsua.

Bug Fixes* Fixed bug in SIP parser: clone_data member of message body is not set.

• Added checking for clock rate mismatch in PJMEDIA master port.
• fixed many memory and handle leaks (there is no single leak now!)
• fixed SIP message leak when transport immediately fails to send the message
• Fixed sound device ID bug: application was unable to use device ID zero because this caused pasound.c to autodetect default device
• Removed duplicated stun.o in pjlib-util Makefile
• Fixed overflow bug in conference.c causing cracking noise (thanks Norman Franke)
• Removed multiple Via headers checking in transport, as proxies will receive multiple Via headers.
• Fixed bug in log length calculation (null character was included) [thanks Klaus Darilion]
• Fixed missing pj_snd_get_info() in nullsound.c
• Changed pasound.c to handle situation when NULL is passed as input/output buffer by PortAudio
• Fixed typos in pj_int_t's comments [thanks Tomasz Ostrowski]
• Removed -DNDEBUG from default CFLAGS [thanks Olivier Gournet]
• Fixed compilation errors when threading is disabled (PJ_HAS_THREADS=0) and safe ioqueue unregistration is disabled (PJ_IOQUEUE_HAS_SAFE_UNREG=0).
• Fixed compilation error in tone generator when using fixed point.
• and many others

For more details about the ticket list, please see Release Notes