http://trac.pjsip.org/repos/milestone/release-1.4

Release Focus
=============

Release focus:

    * Nokia VAS and VAS-Direct Support
    * Initial porting for Symbian 5th Edition
    * Support for SIP Session Timer extension (RFC 4028)
    * New test applications for testing audio subsystem and media handling 

Incompatibility Info
====================

Changes that may break compatibility with previous versions.

    * ticket #930 changes the semantic of various SIP contact parameters/settings:
          o the contact_param field of pjsua_acc_config is now used to represent additional Contact header parameters (not Contact URI parameters).
          o the contact argument in pjsip_dlg_create_uac(), pjsip_dlg_create_uas(), and pjsip_endpt_create_request() now represents the value of the Contact header rather than the Contact URI. This may break the compatibility if application specifies the value without the brackets, for example, "sip:10.0.0.1;transport=udp" instead of "<sip:10.0.0.1;transport=udp>", due to the treatment of the parameters in the Contact header. In the former example, the "transport=udp" parameter will be treated as header parameter and not URI parameter. 

Deprecation Info
================

Info about deprecated APIs:

    * ticket #866 allows application to specify more than one STUN servers in PJSUA-LIB. New settings stun_srv_cnt and stun_srv array are introduced in pjsua_config, and the existing stun_host and stun_domain settings are now deprecated. Backward compatibility is still maintained, if stun_srv_cnt is zero then the values of stun_host and stun_domain will be used. 

List of Enhancements
====================
common

#918
    Initial porting for Symbian 5th edition

pjlib

#921
    New logging option/flag to include caller thread ID
#922
    Option to enable mutex related logging to assist troubleshooting concurrency problems
#935
    New PJLIB API pj_sockaddr_parse2() to parse "HOSTPART[:PORT]" format into the correspondong hostpart and port

pjlib-util
pjnath
pjmedia

#923
    New API to retrieve current jitter buffer state from a stream/session
#929
    Improvement in packet lost concealment (PLC) when handling burst of lost packets

pjmedia-audiodev

#909
    Integration of VoIP Audio Service (VAS) for Nokia S60

pjsip

#833
    Support for SIP Session Timer (RFC 4028)

pjsua-lib

#866
    Allow application to specify more than one STUN servers for more robustness, and continue application startup if STUN resolution fails
#910
    Configurable passthrough codecs based on audio device encoding formats capability.
#912
    Flags in logging configuration to append log file instead of overwriting it
#930
    New PJSUA-LIB account option to add user defined parameters to the Contact header

applications

#920
    New pjsystest application for testing target system/device
#925
    New application to simulate network and system impairments to see how it affects the audio quality

unit-tests

#924
    Loop media transport now allows more than one streams to receive the reflected packets

third-party


List of Bugs
=============

common

#886
    Broken exception in Symbian, potentially will cause undefined behavior when receiving bad SIP message
#915
    Miscellaneous fixes

pjlib

#913
    Concurrency problem in select ioqueue may corrupt descriptor set
#931
    Logging function may infinitely recursively calls itself on Windows Mobile (thanks Emil Sturniolo for the report)
#939
    Throwing exception inside exception handler will cause infinite loop (thanks Roman Puls for the report)
#946
    Symbian kern-exec 0 in resolving IPv6 address.

pjlib-util
pjnath

#916
    Crash in TURN client when TCP connection is used

pjmedia

#783
    Latency issue with Windows Mobile (thanks Johan Lantz for the report)
#919
    An iLBC session must use same mode in both directions.
#926
    SDP parser compliance with RFC 4566 (thanks Johan Lantz for the report)
#942
    MIPS test fails on iLBC encode/decode.
#943
    Assertion in destroying stream port with SRTP in MIPS test
#947
    SRTP stops functioning after the library is restarted

pjmedia-audiodev
pjsip

#877
    Memory consumption of the invite session grows indefinitely if call is running for long period of time and with many re-INVITES
#906
    Transaction is not destroyed when transport timeout event comes later than transaction timeout (thanks Norma Steveley for the report)
#911
    Crash when receiving NOTIFY after subscription is terminated (thanks Johan Lantz for the report)
#927
    PIDF timestamp is not added to the tuple (thanks Johan Lantz for the report)
#934
    Incoming request requiring non-built-in SIP extensions always gets rejected
#938
    Presence PIDF document may be rejected by presence servers that implement strict XML checking (thanks Johan Lantz for the fix)
#948
    Replaces extension stops functioning after the library is restarted.

pjsua-lib

#933
    Incoming OPTIONS may trigger assertion if it arrives when PJSUA-LIB is being shutdown (thanks Johan Lantz for the report)
#945
    Account config may not get initialized with default values if pjsua_acc_config_default() is called before pjsua_init()

applications
unit-tests

#944
    Miscellaneous fixes to pjmedia_test

third-party

#928
    Error linking PJSIP due to inclusion of aes_tables.c in libsrtp (thanks Johan Lantz for the report)