Changelog for libpjsip-devel-2.8.ring-10.2.x86_64.rpm :

* Sun Sep 23 2018 Ákos Szőts - 2.8.0
* common
* Print IPv6 addresses with brackets in the log
* Add SDP attribute SSRC and CNAME
* pjlib, pjlib-util
* Add support for GnuTLS
* Fix build error when building with LibreSSL as SSL backend
* Prevent crash due to access to an already destroyed atomic object
* Replace DNS resolver mutex with group lock
* pjmedia, pjmedia-audiodev
* More clever RTP transport remote address switch
* Optimization: Improve conference mix loop performance
* Add compile time option to enable/disable simple AGC in conference
* Enable wav playlist to play WAV files with extra chunks after DATA chunk
* Support for RTP and RTCP multiplexing
* Support receiving Opus packets with various frame lengths
* Start read operation in UDP media transport in pjmedia_transport_media_start()
* Green screen in the beginning of video call
* Add compile-time setting to specify DTMF duration in ms
* Implement conference signal level adjustment for a specific connection
* Implement RTCP Feedback
* pjsip, pjsua-lib
* Allow to use binary certificate in TLS transport
* Support DTMF via SIP INFO
* New PJSUA API to register a transport factory
* Add more documentation throughout PJSIP to prevent stack buffer overflow
* Update pjsip_resolve() to be able to return more than one resolved address
* New PJSUA & PJSUA2 APIs for instantiating extra audio device
* Revisit IPv4/IPv6 settings and behavior in pjsua
* API for updating remote target via re-INVITE/UPDATE
* Move SRTP setting in PJSUA and PJSUA2 to account setting
* Don\'t raise assert when receiving an incoming call without a pjsua account
* Follow SDP answer changes in 18x & 2xx responses
* Add feature to allow responding incoming INVITE/re-INVITE asynchronously and set the SDP answer
* Updated account matching algo for incoming request
* Skip IPv4 STUN resolution if account is using NAT64
* Add TCP initial receive timeout for server connection
* pjsua2, swig
* Add outbound proxy settings in pjsua2
* applications, python, unit-tests, third-party
* Add C# binding using SWIG, and support for Xamarin.
* common
* Miscellaneous fixes
* Fix various linker error when building as dll on Visual Studio 2015
* pjlib, pjlib-util
* On iOS11, replace_udp_sock() might fail and lead to unusable UDP transport
* SSL connection suddenly gets closed after sending packets intensively
* Initialization of ephemeral ECDH (EECDH) when accepting TLS session works incorrectly when linked with OpenSSL 1.1.0x
* Crash when PJ_GRP_LOCK_DEBUG is set
* Timestamp clock issue when device is asleep in iOS
* pjnath
* Increase default ICE password length as mandated by the RFC
* Missing IPv6 ICE candidates when IPv6 media is configured in PJSUA
* pjmedia, pjmedia-audiodev
* Opus decode/recovery issue when FEC or PLC is enabled
* Crash when receiving SDP with invalid fmtp attribute
* Crash when parsing SDP with an invalid media format description
* Various updates in DTLS-SRTP
* Fixed SID counter for AMR-WB
* Fix incorrect DTMF duration/timestamp for codecs with RTP timestamp unit not using actual sampling rate
* Reset VideoToolbox on iOS when app switches from background to active
* Possible insufficient stream buffer size when using Opus
* Incorrect Opus fmtp settings
* Fix potentially incorrect buffer allocation for video port renderer
* pjmedia-videodev
* Fail to start video preview on Android due to error creating converter
* pjsip, pjsua-lib
* Prevent releasing unacquired lock in SIP dialog
* Unable to destroy certain PJSIP transports
* Fix return code in pjsip_find_msg()
* SDP ignored in 180/183 response without To tag
* on_call_transfer_status() callback is not called when REFER is responded with failure response
* Blocking select() on Android
* Call disconnection in failover scenario due to transport error on previous INVITE request
* Crash in pjsip due to race condition in account\'s keep alive timer
* Via header mismatch in CANCEL
* Fixed crash when transaction timer callback is called after transaction is destroyed
* Prevent double free on Failed STUN resolution
* Fixed RTP socket to bind to any available port if port is zero
* Deadlock between PJSUA LOCK and conference mutex
* Crash in SIP session timer after call hold responded with 422
* Fixed crash when hanging up call if call invite hasn\'t been created
* Re-INVITE not sent for non-registering accounts on IP change
* Race condition in 183 re transmission can result in a deadlock
* Cannot query stream info from pjsua on_stream_created() callback
* Don\'t rearrange media when sending re-INVITE with PJSUA_CALL_REINIT_MEDIA
* pjsua2, swig
* Cannot change active sound device using PJSUA2 setPlaybackDev/setCaptureDev()
* Fixed assertion when setting audio dev in PJSUA2
* Crash when deleting PJSUA2 Account
* SWIG exception in mapping an invalid C++ enum value to Java
* applications, python, unit-tests, third-party
* Update libyuv version to fix linker error when building dll on Visual Studio 2015
* iLBC using memcpy instead of memmove for overlapping mem
* Various PJSUA tests (Python scripts, unit tests) updates and fixes
* List of Tasks
* Review pjsua app sample about pjsua_call_info usage
* Remove deprecated Linux kernel implementation
* Tue Apr 17 2018 szotsakiAATTgmail.com2.7.0
* NAT64 support for IPv4 interoperability
* Linking errors with OpenSSL 1.1.0 when backward compatibility settings turned off
* Android build fail when using NDK r14 caused by the removal of android_alarm.h
* Improve error handling in OpenSSL socket
* Compile time setting for QoS using IP_TOS/IPV6_TCLASS on Darwin OS
* Support ALSA audio device volume setting
* Support DTLS for SRTP keying
* Video Toolbox H264 encoder and decoder for Mac and iOS
* Add option to for the SDP version to not increment when there\'s no change from previous answer/offer
* Support for bcg729
* Add compile-time config for L16 codec
* Add API pjsip_transport_shutdown2() to immediately disconnect a transport
* Add API pjsip_evsub_set_uas_timeout()
* Add API pjsip_multipart_get_raw() to get raw body of a multipart message body
* Add on_rx_offer2() callback for SIP invite
* API to handle IP address change
* Implement CodecParam class in PJSUA2 API as a wrapper for pjmedia_codec_param
* Add setting to retry timer upon transport disconnection failure (503)
* Add multicast option in streamutil sample app
* Add Python 3 support using PJSUA2 API
* List of Bugs
* Failure in configure-android when specifying --use-ndk-cflags with Android NDK r13 or later
* Miscellaneous fixes
* Prevent overflow on pj_generate_unique_string() for android
* Conflict with \"isblank\" when building using g++ 5.4.0
* pj_hash_calc_tolower() might return a different hash value
* Fixed crash due to uncancelled timer if there\'s an error in resolver\'s query transmit
* Assertion in pj_gethostip() when system hostname is empty
* Memory corruption caused by pj_sockaddr_parse()
* iOS specific issue: Error 488 when answering call after app goes background
* Crash in TURN server resolution callback when ICE objects already destroyed
* Fixed crash on pjnath-test due to access to an invalid callback
* Prevent crash when timer refresh with SRTP is interrupted by a re-INVITE
* When receiving an SDP answer for SRTP, process the tag correctly based on the offer
* ICE must use regular nomination when communicating with lite implementations
* Make sure transport SRTP buf size is sufficient before calling srtp_protect() and srtp_protect_rtcp()
* IPv6 media failed if only one of the party uses ICE
* Via-Header mismatch in CANCEL
* Deadlock between dialog lock and transaction group lock
* Deadlock between PJSUA LOCK, transaction group lock, and UA mutex
* Crash when hanging up call if video capture device fails to open
* Possible crash when using session timer due to the early release of dialog pool
* Failure in initializing registration due to unescaped user part in account contact
* Prevent memory leak when rejecting a call from on_incoming_call() callback
* Buffer overrun in PJSIP transaction layer
* Incorrect parsing of zero length multipart body parts
* Crash on pjsip_dlg_create_uac() when specifying URI with valueless header parameter
* Fixed crash in pjsua_destroy if there\'s pending outgoing TCP/TLS transmission
* Cannot send UPDATE when call is ringing
* SRTP error in sending video RTP after hold and unhold
* Fixed crash when accessing video device info in pjsua2
* Crash in getting call info with long Contact header
* List of Tasks
* Update bundled libSRTP version to enable AES-GCM on OpenSSL 1.1.0 or later
* Update libyuv version to fix compile errors on old gcc versions 2.7.1
* Enabling AES-GCM when using external libSRTP version 1.x and 2.0.0
* Update pjsua_get_snd_dev() info before calling on_snd_dev_operation() callback
* Try to allocate larger buffer size instead of immediately returning error when converting pjsip_hdr to SipHeader
* Implement callback wrapper for on_buddy_evsub_state() on pjsua2
* Add compile time option to disable sleep in sip endpoint\'s handle events on ioqueue polling\'s error
* List of Bugs
* Miscellaneous fixes
* Use ar/ranlib from android ndk binutils when building using clang with --use-ndk-cflags option
* When set CXXCFLAGS manually, make sure it is applied correctly.
* Cannot register ioqueue key after double key unregistration
* Add validity checking for numeric header values
* ICE: Use STUN FINGERPRINT attribute when sending keepalives
* Add option for pjsua callback on_stream_created to destroy application\'s supplied media port 2.7.2
* Crash when receiving SDP with invalid fmtp attribute
* Crash when parsing SDP with an invalid media format description
* Mon Jun 26 2017 szotsakiAATTgmail.comPJSIP 2.6
* Windows 10 / Universal Windows Platform port
* Review iOS 10 integration to PJSIP
* pj_pool_safe_release() API
* QoS for IPv6 for platform that supports IPV6_TCLASS
* QoS for darwin OS which supports SO_NET_SERVICE_TYPE
* Add support to parse address string with scope ID
* Implement pj_strtok()/pj_strtok2() to replace strtok()
* Option to regularly send video keyframe in the beginning of video call session
* AES-GCM crypto support for SRTP
* Support for OpenH264 v1.6.0 codec
* Support for setting audio input source capability in Android JNI audio device
* Add function pjmedia_rtp_decode_rtp2()
* Add attach2() and pjmedia_transport_attach2() to pjmedia transport interface
* Add function to get RTP session from stream/vid stream
* Support video window manipulation for native preview
* Add callback to configure SRTP setting and key in pjsua/pjsua2
* Add support to specify Contact params specific to REGISTER requests
* Add function pjsip_tdata_get_dlg()
* Add support to select elliptic curve and signature algorithm for TLS
* Support to generate a synthesized IPv6 address from IPv4 address
* Add option to reinitialize call media transports
* Add option to update call Via address
* Export SIP transport TLS state and TLS certificate info to PJSUA2
* Add WebRTC to third party component
* Fix unused-variable warnings when using -NDEBUG build option
* Failure in configure-android when specifying --use-ndk-cflags with Android NDK r11 or later
* Fail to create resolver when library built with IPv6 but run on system without IPv6
* Timer not fired due to timestamp clock issue in Android
* Crash on using an already destroyed SSL socket
* Various fixes for DNS, primarily for IPv6
* Remove the implementation of PJ_HASH_USE_OWN_TOLOWER
* Fixed ICE stagnation when connectivity check fails
* ICE initialization issues when creating a component/candidate fails.
* Call fails to answer due to ICE media transport init blocking
* Update RTP sequence number during keep-alive
* Modify async dispatch to synchronous on Mac and iOS video device implementation
* Assertion in deinitializing client auth session when dialog creation fails
* Fail to start media due to mismatch address type in SDP connection line
* Assertions in DNS SRV resolution with IPv6 TCP/TLS target
* Add reference counter to pjsip_inv_session to avoid race condition
* Premature STUN socket destruction when there\'s an error during STUN server resolution
* Enable IPv6 in ICE transport/TURN in PJSUA
* Assertion if remote removes some media lines in previous SDP negotiations
* Assertion when session timer is disabled and PJSIP receives 422
* Escape \'AATT\' in the Replaces parameter of REFER request
* Assertion in re-INVITE with PJSUA_CALL_REINIT_MEDIA
* Crash on UDP transport restart
* Support OpenSSL 1.1.0
* Support for Android NDK r12 and Android N
* Migrate Android projects from Eclipse to Android Studio
* Thu Oct 20 2016 szotsakiAATTgmail.comPJSIP 2.5.5- List of Enhancements-- common:--- #1881 Visual Studio 2015 support-- pjlib, pjlib-util:--- #1894 Improve ioqueue performance on multithreadeded environment--- #1909 GUID implementation for Android-- pjmedia, pjmedia-audiodev:--- #1847 Upgrade libsrtp version and support for AES-256 crypto--- #1888 Support for WebRTC Acoustic Echo Cancellation--- #1896 Update default audio device backends--- #1897 Support Ffmpeg 2.8--- #1904 Support for Opus codec--- #1907 Remove pjmedia
* circular dependency-- pjsip, pjsua-lib:--- #1892 Add pjsua/pjsua2 callback to notify incoming re-INVITE without offer--- #1908 Support opening speaker only in pjsua/pjsua2--- #1913 Add callback for address change notification from STUN keep alive--- #1914 Ignore STUN error after pjstun_get_mapped_addr2()--- #1915 Add API pjsip_udp_transport_start2()-- pjsua2, swig:--- #1879 Set Video Codec Param using PJSUA2 API- List of Bugs-- common:--- #1882 Miscellaneous fixes-- pjlib, pjlib-util:--- #1889 DNS parser returns error on parsing RR type OPT--- #1912 Build error on Alpine linux (musl libc)-- pjnath:--- #1891 ICE negotiation fails after each component has successful connectivity check--- #1903 Crash when cleaning STUN response cache-- pjmedia, pjmedia-audiodev:--- #1884 Audio stutter on remote side after disconnecting stream from microphone in the conference bridge--- #1887 No output frame returned by iLBC encoder on iOS when using multiple frames per packet-- pjmedia-videodev:--- #1880 Incorrect orientation after switching video capture or when using back camera-- pjsip, pjsua-lib:--- #1311 Locking account to specific TCP/TLS listener will cause registration loop (thanks Tony Million for the report)--- #1883 Crash in decrementing transport reference count--- #1885 Race conditions in event subscription--- #1886 Fixed destruction of locked mutex in SIP dialog--- #1893 iOS application getting killed after pjsua fail to release a disconnected transport--- #1895 Terminate subscription when receiving non 2xx Notify response without Retry-After header--- #1898 Transport may never gets destroyed when connected event comes while transport is shutting down--- #1899 Create stream for inactive media to allow keep-alive and RTCP traffics--- #1901 Fix crash when async_cnt is set to a value greater than one for SIP TLS transport--- #1902 Crash when endpoint has multiple worker threads and SIP TCP transport is disconnected during incoming call handling--- #1905 Fixed assertion in call redirection-- pjsua2, swig:--- #1911 Callback onCallSdpCreated() (pjsua2) doesn\'t modify the SDP--- #1916 onCreateMediaTransport() callback might not be called on PJSUA2- List of Tasks-- #1841 Review Android audio output latency-- #1906 Remove PortAudio source from package
* Wed Jun 01 2016 szotsakiAATTgmail.com- Initial version of Ring specific PJSIP version